Internet Draft Alan Johnston Document: draft-johnston-sip-call-flows-00.txt Steve Donovan Category: Informational Robert Sparks Chris Cunningham Kevin Summers MCI WorldCom October 1999 SIP Telephony Call Flow Examples Status of this Memo This document is an Internet-Draft and is in full conformance with all provisions of Section 10 of RFC2026[1]. Internet-Drafts are working documents of the Internet Engineering Task Force (IETF), its areas, and its working groups. Note that other groups may also distribute working documents as Internet-Drafts. Internet-Drafts are draft documents valid for a maximum of six months and may be updated, replaced, or obsoleted by other documents at any time. It is inappropriate to use Internet- Drafts as reference material or to cite them other than as "work in progress." The list of current Internet-Drafts can be accessed at http://www.ietf.org/ietf/1id-abstracts.txt The list of Internet-Draft Shadow Directories can be accessed at http://www.ietf.org/shadow.html. Abstract This document gives examples of SIP IP Telephony call flows. Elements in these call flows include SIP User Agents and Clients, SIP Proxy and Redirect Servers, and Gateways to the PSTN (Public Switch Telephone Network). Scenarios include SIP Registration, SIP to SIP calling, SIP to Gateway, Gateway to SIP, and Gateway to Gateway via SIP. Call flow diagrams and message details are shown. PSTN telephony protocols are illustrated using SS7 (Signaling System 7), ISDN (Integrated Services Digital Network) and FGB (Feature Group B) circuit associated signaling. PSTN calls are illustrated using global telephone numbers from the PSTN and from private extensions served on a PBX (Private Branch Exchange). Johnston, et al. [Page 1] Internet Draft SIP Telephony Call Flow Examples October, 1999 Table of Contents 1 Overview...................................................3 1.1 General Assumptions........................................3 1.2 Legend for Message Flows...................................5 1.3 SIP Protocol Assumptions...................................6 2 SIP Registration Services..................................7 2.1 Success Scenarios..........................................7 2.1.1 SIP Client New Registration................................7 2.1.2 User Cancels Registration..................................9 2.1.3 User updates contact list.................................10 2.1.4 User Requests Current Contact List........................12 2.2 Failure Scenarios.........................................13 2.2.1 Unsuccessful SIP registration.............................13 3 SIP to SIP Dialing........................................15 3.1 Success Scenarios.........................................15 3.1.1 Successful SIP to SIP through two proxies.................16 3.1.2 Successful SIP to SIP with Proxy failure..................26 3.1.3 Successful SIP to SIP through SIP Firewall Proxy..........34 3.1.4 Successful SIP to SIP via Redirect and Proxy..............43 3.2 Failure Scenarios.........................................50 3.2.1 Unsuccessful SIP to SIP no answer.........................50 3.2.2 Unsuccessful SIP to SIP busy..............................56 3.2.3 Unsuccessful SIP to SIP no response.......................60 3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable...........65 4 SIP to Gateway Dialing....................................71 4.1 Success Scenarios.........................................71 4.1.1 Successful SIP to ISUP PSTN call..........................72 4.1.2 Successful SIP to ISDN PBX call...........................79 4.1.3 Successful SIP to ISUP PSTN call with overflow............87 4.2 Failure Scenarios.........................................95 4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN........96 4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN..........101 4.2.3 Unsuccessful SIP to PSTN: ANM Timeout....................105 5 Gateway to SIP Dialing...................................111 5.1 Success Scenarios........................................111 5.1.1 Successful PSTN to SIP call..............................112 5.1.2 Successful PSTN to SIP call, Fast Answer.................118 5.1.3 Successful PBX to SIP call...............................123 5.2 Failure Scenarios........................................128 5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL....128 5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL.....130 5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones..134 5.2.4 Unsuccessful PSTN->SIP, ACM timeout......................139 5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless SPS.......144 5.2.6 Unsuccessful PSTN->SIP, ANM timeout......................150 6 Gateway to Gateway Dialing via SIP Network...............155 6.1 Success Scenarios........................................155 6.1.1 Successful ISUP PSTN to ISUP PSTN call...................156 6.1.2 Successful FGB PBX to ISDN PBX call with overflow........163 Johnston, et al. Informational [Page 2] Internet Draft SIP Telephony Call Flow Examples October, 1999 7 Acknowledgements.........................................172 8 References...............................................172 1 Overview The call flows shown in this document were developed in the design of a carrier-class SIP IP Telephony network. They represent an example minimum set of functionality for SIP to be used in IP Telephony applications. It is the hope of the authors that this document will be useful for SIP implementors, designers, and protocol researchers alike and will help further the goal of a standard SIP implementation for IP Telephony. It is envisioned that as changes to the standard and additional RFCs are added that this document will reflect those changes and represent the current state of a standard interoperable SIP IP Telephony implementation. These call flows are based on the current version 2.0 of SIP in RFC2543[2]. Additions and changes to SIP necessary for PSTN interworking are referenced as IETF Internet-Drafts as they are used in the call flows. Various PSTN signaling protocols are illustrated in this document: SS7 (Signaling System 7), ISDN (Integrated Services Digital Network), and FGB (Feature Group B) circuit associated signaling. They were chosen to illustrate the nature of SIP/PSTN interworking _ they are not a complete or even representative set. Also, some details and parameters of these PSTN protocols have been omitted. The intent of this document was not to provide a complete and exact mapping of PSTN protocols to SIP. Rather the emphasis is on the SIP signaling, the message interaction, and the modifications to SIP currently proposed to solve IP Telephony issues. Finally, these call flows show a minimal implementation. Not even basic telephony features such as call forwarding or call waiting are included. A typical carrier-class implementation of a basic set of telephony features using SIP is described in another document[3]. 1.1 General Assumptions A number of architecture, network, and protocol assumptions underlie the call flows in this document. They are outlined in this section so that they may be taken into consideration. Differences in these assumptions will affect the nature of the call flows. The authentication of SIP User Agents in these example call flows is performed using SIP Digest. No authentication of Gateways is shown, since it is assumed that: Johnston, et al. Informational [Page 3] Internet Draft SIP Telephony Call Flow Examples October, 1999 . Gateways will only accept calls routed through a trusted Proxy. . Proxies will perform the Client authentication. . The Proxy and the Gateway will authenticate each other using IPSec[4]. The SIP Proxy Server has access to a Location Manager and other databases. Information present in the Request-URI and the context (From header) is sufficient to determine to which proxy or gateway the message should be routed. In most cases, a primary and secondary route will be determined in case of Proxy or Gateway failure downstream. The Proxy Servers in these call flows insert Record-Route headers into requests to ensure that they are in the signaling path for future message exchanges. Gateways receive enough information in the Request-URI field to determine how to route a call, i.e. what trunk group or link to select, what digits to outpulse, etc. Gateways provide tones (ringing, busy, etc) and announcements to the PSTN side based on SIP response messages, or pass along audio in-band tones (ringing, busy tone, etc.) in an early media stream to the SIP side. Two types of Gateways are described in this document: . Network Gateway. This high port count PSTN gateway originates and terminates calls to the PSTN. It's use is shared by many customers. Incoming calls from the PSTN have the From header populated with a SIP URL containing the telephone number from the calling party telephone number, if available. A Network Gateway typically uses carrier protocols such as SS7. . Enterprise Gateway. This low port count PBX (Private Branch Exchange) gateway has trunks or lines for a single customer or user. Incoming calls from the PBX have the From header populated with a provisionable string which uniquely identifies the customer, trunk group, or carrier. This allows private numbers to be interpreted in their correct context. An Enterprise Gateway typically uses SS7, ISDN, circuit associated signaling, or other PBX interfaces. The interactions between the Proxy and Gateway can be summarized as follows: . The SIP Proxy Server performs digit analysis and lookup and locates the correct gateway. Johnston, et al. Informational [Page 4] Internet Draft SIP Telephony Call Flow Examples October, 1999 . The SIP Proxy Server performs gateway location based on primary and secondary routing. Digit handling by the Gateways will be as follows: . Dialed digits received from a Network or Enterprise Gateway will be put in a SIP URL with a telephone number. The number will either be globalized (e.g. sip:+1-314-555-1111@ngw.wcom.com ;user=phone) or left as a private number (sip:555-6666,phone- context=p1234@gw.wcom.com;user=phone) which will require interpretation based on From header. The "phone-context=" qualifier is used to interpret the private number. It is used the same as the tag of the same name from the Tel URL draft[5]. However, its use in the user portion of the SIP URL should not require changes to parsers. All Gateways will need to be provisioned to be able to parse the user portion of a Request- URI to determine the customer, trunk group, or circuit referenced. . The From header will be populated with a SIP URL with a telephone number if it is Calling Party number (CgPN) from the PSTN. If it is an Enterprise Gateway, a provisionable string which uniquely identifies the customer, trunk group, or carrier will be used in the sip URI (e.g. From: sip:ProvisionableString@gw1.wcom.com ;user=phone). . Note that an alternative to using a SIP URL for telephone numbers is the TEL URL[5]. The major difference between using the SIP URL and the TEL URL is that the SIP URL is routable in a SIP network (resolves down to an IP address) where the TEL URL is not (it just represents digits). For example, a SIP URL can be used in a Contact header, but a TEL URL can not. These flows show UDP for transport. TCP could also be used. 1.2 Legend for Message Flows Important Note: In this text version of this Internet Draft, figures containing the message flows have been deleted. They will be present in the next draft of this document. A PostScript (.ps) and Acrobat (.pdf) version are available which contain the figures. Dashed lines _ represent control messages that are mandatory to the call scenario. These control messages can be SIP or PSTN signaling. Solid lines _ represent media paths between network elements. Dashed line with parenthesis around name - represent optional control messages. Johnston, et al. Informational [Page 5] Internet Draft SIP Telephony Call Flow Examples October, 1999 Messages are identified in the Figures as F1, F2, etc. This references the message details in the table that follows the Figure. Comments in the message details are shown in the following form: /* Comments. */ 1.3 SIP Protocol Assumptions Except for the following, this call flows document uses the April 1999 version 2.0 of SIP defined by RFC2543[2]. The following changes/extensions are assumed throughout: . A Contact header is included with every INVITE message. . A Contact header is included in every 200 OK Response. . The 183 Session Progress response message[5] is used in SIP to Gateway and Gateway to Gateway via SIP calling (Sections 4 and 6). The 183 response with SDP will cause the User Agent to immediately play the SDP media stream to hear in-band call progress information. See Section 4 for more information. . A Content-Length header is present in every message, set to zero if there is no message body. . The final entry in a Route header is always the Contact information obtained from the INVITE or 200 OK messages. . In the SDP message bodies, the time field is "t=0 0" It is expected that an actual SDP message body would have a non-zero start timestamp. Johnston, et al. Informational [Page 6] Internet Draft SIP Telephony Call Flow Examples October, 1999 2 SIP Registration Services 2.1 Success Scenarios Registration either validates or invalidates a SIP client for user services provided by the SIP proxy and/or SIP server. Additionally, the client provides one or more contact locations to the SIP server with the registration request. 2.1.1 SIP Client New Registration User B initiates a new SIP session with the SIP server (i.e. the user "logs on to" the SIP server). User B sends a SIP REGISTER request to the SIP server. The request includes the user's contact list. The SIP server provides a challenge to User B. User B enters her/his valid user ID and password. User B's SIP client encrypts the user information according to the challenge issued by the SIP server and sends the response to the SIP server. The SIP server validates the user's credentials. It registers the user in its contact database and returns a response (200 OK) to User B's SIP client. The response includes the user's current contact list in Contact headers. The format of the authentication shown is SIP digest as described by RFC2543[2]. Message Details REGISTER F1 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456789@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Content-Length: 0 Unauthorized F2 SIP server-> User B SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy Johnston, et al. Informational [Page 7] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy Call-ID: 123456789@here.com CSeq: 1 REGISTER WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", algorithm="MD5" Content-Length: 0 REGISTER F3 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456790@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Length: 0 200 OK F4 SIP server -> B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 1234567890@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Content-Length: 0 Johnston, et al. Informational [Page 8] Internet Draft SIP Telephony Call Flow Examples October, 1999 2.1.2 User Cancels Registration User B wishes to cancel her/his registration with the SIP registrar/redirect server. User B sends a SIP REGISTER request to the SIP server. The request has an expiration period of 0 and applies to all existing contact locations. Since the user already has authenticated with the server, the user supplies authentication credentials with the request and is not challenged by the server. The SIP server validates the user's credentials. It clears the user's contact list, and returns a response (200 OK) to User B's SIP client. Message Details REGISTER F1 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456791@here.com CSeq: 1 REGISTER Expires: 0 Contact: * Authorization:Digest username="UserB", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Length: 0 200 OK F2 SIP server -> B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 1234567891@here.com CSeq: 1 REGISTER Content-Length: 0 Johnston, et al. Informational [Page 9] Internet Draft SIP Telephony Call Flow Examples October, 1999 2.1.3 User updates contact list User B wishes to update the list of addresses where the SIP server will redirect INVITE requests. Note this scenario assumes that Scenario 2.1.1 has taken place, but 2.1.2 has not (i.e. User B currently has 3 contacts registered with SS2.) User B sends a SIP REGISTER request to the SIP server. User B's request includes an updated contact list. Since the user already has authenticated with the server, the user supplies authentication credentials with the request and is not challenged by the server. The SIP server validates the user's credentials. It registers the user in its contact database, updates the user's contact list, and returns a response (200 OK) to User B's SIP client. The response includes the user's current contact list in Contact headers. Message Details REGISTER F1 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456791@here.com CSeq: 1 REGISTER Contact: mailto:UserB@there.com Authorization:Digest username="UserB", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Length: 0 200 OK F2 SIP server -> B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 1234567891@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Johnston, et al. Informational [Page 10] Internet Draft SIP Telephony Call Flow Examples October, 1999 Contact: tel:+1-972-555-2222 Contact: mailto:UserB@there.com Content-Length: 0 Johnston, et al. Informational [Page 11] Internet Draft SIP Telephony Call Flow Examples October, 1999 2.1.4 User Requests Current Contact List User B sends a register request to the Proxy serverer containing no Contact headers, indicating the user wishes to query the server for the user's current contact list. Since the user already has authenticated with the server, the user supplies authentication credentials with the request and is not challenged by the server. The SIP server validates the user's credentials. It registers the user in its contact database and returns a response (200 OK) to User B's SIP client. The response includes the user's current contact list in Contact headers. Message Details REGISTER F1 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456792@here.com CSeq: 1 REGISTER Authorization:Digest username="UserB", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Length: 0 200 OK F2 SIP server -> B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 1234567892@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Contact: mailto:UserB@there.com Content-Length: 0 Johnston, et al. Informational [Page 12] Internet Draft SIP Telephony Call Flow Examples October, 1999 2.2 Failure Scenarios 2.2.1 Unsuccessful SIP registration User B sends a SIP REGISTER request to the SIP server. The SIP server provides a challenge to User B. User B enters her/his user ID and password. User B's SIP client encrypts the user information according to the challenge issued by the SIP server and sends the response to the SIP server. The SIP server attempts to validate the user's credentials, but they are not valid (the user's password does not match the password established for the user's account). The server returns a response (401 Unauthorized) to User B's SIP client. Message Details REGISTER F1 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456789@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Content-Length: 0 Unauthorized F2 SIP server-> User B SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456789@here.com Johnston, et al. Informational [Page 13] Internet Draft SIP Telephony Call Flow Examples October, 1999 CSeq: 1 REGISTER WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", algorithm="MD5" Content-Length: 0 REGISTER F3 B->SIP server REGISTER sip:ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 123456791@here.com CSeq: 1 REGISTER Contact: TheLittleGuy Contact: sip:+1-972-555-2222@gw1.wcom.com;user=phone Contact: tel:+1-972-555-2222 Authorization:Digest username="UserB", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Length: 0 Note: The response above encodes the incorrect password _IForgot_ Unauthorized F4 SIP server-> User B SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy To: TheLittleGuy Call-ID: 1234567891@here.com CSeq: 1 REGISTER WWW-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", algorithm="MD5" Content-Length: 0 Johnston, et al. Informational [Page 14] Internet Draft SIP Telephony Call Flow Examples October, 1999 3 SIP to SIP Dialing 3.1 Success Scenarios This section details calls between two SIP User Agent Clients (UACs) _ User A and User B. User A (TheLittleGuy sip:UserA@here.com) and User B (TheBigGuy sip:UserB@there.com) are assumed to be SIP phones or SIP-enabled devices. Calls route using at least one SIP Proxy server. The successful calls show the initial signaling, the exchange of media information in the form of SDP payloads, the establishment of the media session, then finally the termination of the call. SIP digest authentication is used by the first Proxy Server to authenticate the caller User A. It is assumed that User B has registered with Proxy Server SS2 as per Section 2.1 to be able to receive the calls. Johnston, et al. Informational [Page 15] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.1.1 Successful SIP to SIP through two proxies In this scenario, User A completes a call to User B using two proxies SS1 and SS2. The initial INVITE (F1) does not contain the Authorization credentials SS1 requires, so a 407 Proxy Authorization response is sent containing the challenge information. A new INVITE (F4) is then sent containing the correct credentials and the call proceeds. The call terminates when User B disconnects by initiating a BYE message. SS1 inserts a Record-Route header into the INVITE message to ensure that it is present in all subsequent message exchanges. SS2 also inserts itself into the Record-Route header. The ACK (F15) and BYE (F18) both have a Route header. A tag is inserted by User B in message F9 since the initial INVITE message contains more than one Via header and may have been forked. Message Details INVITE F1 A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 16] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* Proxy 1 challenges User A for authentication */ 407 Proxy Authorization Required F2 Proxy 1 -> User A SIP/2.0 407 Proxy Authorization Required Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Proxy-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", algorithm="MD5" Content-Length: 0 ACK F3 A -> Proxy 1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 /* User A responds be re-sending the INVITE with authentication credentials in it. */ INVITE F4 A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length:132 Johnston, et al. Informational [Page 17] Internet Draft SIP Telephony Call Flow Examples October, 1999 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* Proxy 1 accepts the credentials and forwards the INVITE to Proxy 2. Proxy 1 is assumed to have been authenticated by Proxy 2 using IPSec. Client for A prepares to receive data on port 49170 from the network. */ INVITE F5 Proxy 1 -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F6 Proxy 1 -> A) Johnston, et al. Informational [Page 18] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F7 Proxy 2 ->B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F8 Proxy 2 -> Proxy 1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F9 Johnston, et al. Informational [Page 19] Internet Draft SIP Telephony Call Flow Examples October, 1999 B -> Proxy 2 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringin g F10 Proxy 2 -> Proxy 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F11 Proxy1 ->A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 200 OK F12 B -> Proxy 2 SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 20] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F13 Proxy 2 -> Proxy 1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F14 Proxy 1 -> A Johnston, et al. Informational [Page 21] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F15 A -> Proxy 1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 ACK Content-Length: 0 ACK F16 Proxy 1 -> Proxy 2 ACK sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 ACK Content-Length: 0 ACK F17 Johnston, et al. Informational [Page 22] Internet Draft SIP Telephony Call Flow Examples October, 1999 Proxy 2 ->B ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345601@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B */ /* User B Hangs Up with User A. */ BYE F18 User B -> Proxy 2 BYE sip: UserA@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 Route: , From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 BYE F19 Proxy 2 -> Proxy 1 BYE sip: UserA@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Johnston, et al. Informational [Page 23] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP there.com:5060 Route: From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 BYE F20 Proxy 1 -> User A BYE sip: UserA@here.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 200 OK F21 User A -> Proxy 1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 200 OK F22 Proxy 1 -> Proxy 2 SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 Johnston, et al. Informational [Page 24] Internet Draft SIP Telephony Call Flow Examples October, 1999 200 OK F23 Proxy 2 -> User B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345601@here.com CSeq: 1 BYE Content-Length: 0 Johnston, et al. Informational [Page 25] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.1.2 Successful SIP to SIP with Proxy failure In this scenario, User A completes a call to User B via a Proxy Server. User A is configured for a primary SIP Proxy Server SS1 and a secondary SIP Proxy Server SS2 (Or is able to use DNS SRV records to locate SS1 and SS2). SS1 is out of service and does not respond to INVITEs (it is reachable, but unresponsive). After sending a CANCEL to SS1, User A then completes the call to User B using SS2. Message Details INVITE F1 A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 INVITE F2 A -> Proxy 1 Same as Message F1 INVITE F3 A -> Proxy 1 Same as Message F1 INVITE F4 Johnston, et al. Informational [Page 26] Internet Draft SIP Telephony Call Flow Examples October, 1999 A -> Proxy 1 Same as Message F1 INVITE F5 A -> Proxy 1 Same as Message F1 INVITE F6 A -> Proxy 1 Same as Message F1 INVITE F7 A -> Proxy 1 Same as Message F1 /* User A gives up on the unresponsive proxy and sends a CANCEL. If any 200 OK responses come back to the INVITE, User A sends an ACK, then a BYE. */ CANCEL F8 A -> Proxy 1 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 CANCEL INVITE F9 A -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Johnston, et al. Informational [Page 27] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* Proxy 2 challenges User A for authentication */ 407 Proxy Authorization Required F10 Proxy 2 -> User A SIP/2.0 407 Proxy Authorization Required Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Proxy-Authenticate: Digest realm="MCI WorldCom SIP", domain="wcom.com", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", stale="FALSE", algorithm="MD5" Content-Length: 0 ACK F11 A -> Proxy 2 ACK sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345601@here.com CSeq: 1 INVITE Content-Length: 0 /* User A responds be re-sending the INVITE with authentication credentials in it. */ INVITE F12 A -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Johnston, et al. Informational [Page 28] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345602@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss2.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* Proxy 2 accepts the credentials and forwards the INVITE to User B. Client for A prepares to receive data on port 49170 from the network. */ INVITE F13 Proxy 2 ->B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345602@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 29] Internet Draft SIP Telephony Call Flow Examples October, 1999 (100 Trying F14 Proxy 2 -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345602@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F15 B -> Proxy 2 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F16 Proxy 2 -> A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 INVITE Content-Length: 0 200 OK F17 B -> Proxy 2 SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 INVITE Johnston, et al. Informational [Page 30] Internet Draft SIP Telephony Call Flow Examples October, 1999 Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F18 Proxy 2 ->A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F19 A -> Proxy 2 ACK sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 ACK Content-Length: 0 ACK F20 Proxy 2 ->B Johnston, et al. Informational [Page 31] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345602@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B */ /* User B Hangs Up with User A. */ BYE F21 User B -> Proxy 2 BYE sip: UserA@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 Route: From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345602@here.com CSeq: 1 BYE Content-Length: 0 BYE F22 Proxy 2 -> User A BYE sip: UserA@here.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345602@here.com CSeq: 1 BYE Content-Length: 0 200 OK F23 User A -> Proxy 2 Johnston, et al. Informational [Page 32] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345602@here.com CSeq: 1 BYE Content-Length: 0 200 OK F24 Proxy 2 -> User B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345602@here.com CSeq: 1 BYE Content-Length: 0 Johnston, et al. Informational [Page 33] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.1.3 Successful SIP to SIP through SIP Firewall Proxy User A completes a call to User B through a Firewall Proxy and a SIP Proxy. The signaling message exchange is identical to 3.1.1 but the media stream setup is not end-to-end _ the Firewall proxy terminates both media streams and bridges them. This is done by the Proxy modifying the SDP in the INVITE (F1) and 200 OK (F11) messages. In addition to firewall traversal, this back-to-back User Agent Client and User Agent Server could be used as part of an anonymizer service (in which all identifying information on User A would be removed), or to perform codec media conversion, such as mu-law to A- law conversion of PCM on an international call. Message Details INVITE F1 A->SIP FW INVITE sip:UserB@ fwp1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 34] Internet Draft SIP Telephony Call Flow Examples October, 1999 Client for A prepares to receive data on port 49170 from the network.*/ INVITE F2 SS FW -> SS1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length:134 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 200.201.202.203 m=audio 1000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F3 SIP FW-> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 /* SIP FW prepares to proxy data from port 1000 to 100.101.102.103/49170. SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to User B */ Johnston, et al. Informational [Page 35] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE F4 SS1->B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:134 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 200.201.202.203 m=audio 1000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F5 SS1 -> SIP FW) SIP/2.0 100 Trying Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 (100 Trying F6 B -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 36] Internet Draft SIP Telephony Call Flow Examples October, 1999 180 Ringing F7 B->SS1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F8 SS1 -> SIP FW SIP/2.0 180 Ringing Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F9 SIP FW -> A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 200 OK F10 B->SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy Johnston, et al. Informational [Page 37] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 133 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F11 SS1 -> SIP FW SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a =rtpmap:0 PCMU/8000 200 OK F12 SIP FW -> A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheLittleGuy Johnston, et al. Informational [Page 38] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 200.201.202.203 m=audio 1002 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* The gateway prepares to proxy packets from port 1001 to 110.111.112.113/3456 ACK F13 A->SIP FW ACK sip:UserB@fwp1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 ACK F14 SIP FW -> SS1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 ACK F15 SS1->B ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 39] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and the SIP GW and between the SIP GW and B*/ /* User A Hangs Up with User B. */ BYE F16 A->SIP FW BYE sip: UserB@fwp1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: , From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 BYE F17 SIP FW -> SS1 BYE sip: UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 BYE F18 SS1->B Johnston, et al. Informational [Page 40] Internet Draft SIP Telephony Call Flow Examples October, 1999 BYE sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F19 B->SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F20 SS1 -> SIP FW SIP/2.0 200 OK Via: SIP/2.0/UDP fwp1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F21 SIP FW -> A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 Johnston, et al. Informational [Page 41] Internet Draft SIP Telephony Call Flow Examples October, 1999 Johnston, et al. Informational [Page 42] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.1.4 Successful SIP to SIP via Redirect and Proxy In this scenario, User A places a call to User B using first a Redirect server then a Proxy Server. The INVITE message is first sent to the Redirect Server. The Server returns a 302 Moved Temporarily response (F2) containing a Contact header with User B's current SIP address. User A then generates a new INVITE and sends to User B via the Proxy Server and the call proceeds normally. The call is terminated when User B sends a BYE message. Johnston, et al. Informational [Page 43] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details INVITE F1 A->Redir Proxy INVITE sip:UserB@redirect.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /*Client for A prepares to receive data on port 49170 from the network.*/ 302 Moved Temporarily F2 Redir Proxy ->A SIP/2.0 302 Moved Temporarily Contact: sip:UserB@ss2.wcom.com Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F3 A->Redir Proxy ACK sip:UserB@redirect.wcom.com SIP/2.0 Johnston, et al. Informational [Page 44] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F4 A -> Proxy INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 2 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 INVITE F5 Proxy -> B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 2 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 45] Internet Draft SIP Telephony Call Flow Examples October, 1999 (100 Trying F6 Proxy -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 2 INVITE Content-Length: 0 (100 Trying ) F7 B -> Proxy SIP/2.0 100 Trying Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 2 INVITE Content-Length: 0 180 Ringing F8 B->Proxy SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 INVITE Content-Length: 0 180 Ringing F9 Proxy->A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 INVITE Johnston, et al. Informational [Page 46] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Length: 0 200 OK F10 B->Proxy SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F11 Proxy->A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy >;tag=314159 Call-Id: 12345600@here.com CSeq: 2 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F12 A -> Proxy Johnston, et al. Informational [Page 47] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 ACK Content-Length: 0 ACK F13 Proxy -> B ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 2 ACK Content-Length: 0 /* RTP streams are established between A and B*/ /* User B Hangs Up with User A. */ BYE F14 B->Proxy BYE sip: UserA@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP there.com:5060 Route: From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345600@here.com CSeq: 1 BYE Content-Length: 0 BYE F15 Proxy->A Johnston, et al. Informational [Page 48] Internet Draft SIP Telephony Call Flow Examples October, 1999 BYE sip: UserA@here.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345600@here.com CSeq: 1 BYE Content-Length: 0 200 OK F16 A -> Proxy SIP/2.0 200 OK Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345600@here.com CSeq: 1 BYE Content-Length: 0 200 OK F17 Proxy -> B SIP/2.0 200 OK Via: SIP/2.0/UDP there.com:5060 From: TheLittleGuy ;tag=314159 To: TheBigGuy Call-Id: 12345600@here.com CSeq: 1 BYE Content-Length: 0 Johnston, et al. Informational [Page 49] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.2 Failure Scenarios 3.2.1 Unsuccessful SIP to SIP no answer In this scenario, User A gives up on the call before User B answers (sends a 200 OK response). User A sends a CANCEL (F9) since no final response had been received from User B. If a 200 OK to the INVITE had crossed with the CANCEL, User A would have sent an ACK then a BYE to User B in order to properly terminate the call. Note that the CSeq of the CANCEL message (F9) is not incremented. This is so that downstream clients can match the To, From, Call-ID, and CSeq of the CANCEL to the INVITE to decide which request to terminate. Johnston, et al. Informational [Page 50] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details INVITE F1 A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length:132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /*Client for A prepares to receive data on port 49170 from the network.*/ INVITE F2 Proxy 1 -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length:132 v=0 Johnston, et al. Informational [Page 51] Internet Draft SIP Telephony Call Flow Examples October, 1999 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F3 Proxy 1 -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F4 Proxy 2 ->B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F5 Proxy 2 -> Proxy 1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy Johnston, et al. Informational [Page 52] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F6 B -> Proxy 2 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F7 Proxy 2 -> Proxy 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F8 Proxy1 -> A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 53] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* User A gives up and sends a CANCEL. If a 200 OK reply to the INVITE crossed with the CANCEL and was received by User A, User A would send an ACK then a BYE to terminate the call.*/ CANCEL F9 A -> Proxy 1 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F10 Proxy 1 -> Proxy 2 CANCEL sip: UserA@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F11 Proxy 2 ->B CANCEL sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 200 OK F12 A -> Proxy 1 SIP/2.0 200 OK Johnston, et al. Informational [Page 54] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 200 OK F13 Proxy 1 -> Proxy 2 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 200 OK F14 Proxy 2 ->B SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 Johnston, et al. Informational [Page 55] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.2.2 Unsuccessful SIP to SIP busy In this scenario, User B is busy and sends a 486 Busy Here response to User A's INVITE. The 4xx response is ACKed at each signaling leg. Message Details INVITE F1 User A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /*Client for A prepares to receive data on port 49170 from the network.*/ INVITE F2 Proxy 1 -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com Johnston, et al. Informational [Page 56] Internet Draft SIP Telephony Call Flow Examples October, 1999 CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F3 Proxy 1 -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F4 Proxy 2 -> User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 57] Internet Draft SIP Telephony Call Flow Examples October, 1999 (100 Trying F5 Proxy 2 -> Proxy 1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 486 Busy Here F6 User B -> Proxy 2 SIP/2.0 486 Busy Here Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F7 Proxy 2 -> User B ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 486 Busy Here F8 Proxy 2 -> Proxy 1 SIP/2.0 486 Busy Here Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 58] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK F9 Proxy 1 -> Proxy 2 ACK sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 486 Busy Here F10 Proxy 1 -> User A SIP/2.0 486 Busy Here Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F11 User A -> Proxy 1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 59] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.2.3 Unsuccessful SIP to SIP no response In this example, there is no response from User B to User A's INVITE messages being re-transmitted by Proxy 2. After the sixth re- transmission, Proxy 2 gives up and sends a CANCEL to User B and a 480 No Response to User A. Note that the CANCEL would also be retransmitted six times, as governed by SIP timer T1 as in Section 5.2.6. Message Details INVITE F1 User A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 60] Internet Draft SIP Telephony Call Flow Examples October, 1999 /*Client for A prepares to receive data on port 49170 from the network.*/ INVITE F2 Proxy 1 -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F3 Proxy 1 -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F4 Proxy 2 -> User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Johnston, et al. Informational [Page 61] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F5 Proxy 2 -> Proxy 1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F6 Proxy 2 -> User B Resend of Message F4 INVITE F7 Proxy 2 -> User B Resend of Message F4 INVITE F8 Proxy 2 -> User B Resend of Message F4 INVITE F9 Proxy 2 -> User B Resend of Message F4 INVITE F10 Proxy 2 -> User B Resend of Message F4 INVITE F11 Proxy 2 -> User B Resend of Message F4 CANCEL F12 Proxy 2 -> User B Johnston, et al. Informational [Page 62] Internet Draft SIP Telephony Call Flow Examples October, 1999 CANCEL sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 480 No Response F13 Proxy 2 -> Proxy 1 SIP/2.0 480 No Response Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F14 Proxy 1 -> Proxy 2 ACK sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 480 No Response F15 Proxy 1 -> User A SIP/2.0 480 No Response Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F16 User A -> Proxy 1 Johnston, et al. Informational [Page 63] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 64] Internet Draft SIP Telephony Call Flow Examples October, 1999 3.2.4 Unsuccessful SIP to SIP Temporarily Unavailable In this scenario, User B initially sends a 180 Ringing response to User A, indicating that alerting is taking place. However, then a 480 Unavailable is then sent to User A. This response is acknowledged then proxied back to User A. Johnston, et al. Informational [Page 65] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details INVITE F1 A -> Proxy 1 INVITE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /*Client for A prepares to receive data on port 49170 from the network.*/ INVITE F2 Proxy 1 -> Proxy 2 INVITE sip:UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 Johnston, et al. Informational [Page 66] Internet Draft SIP Telephony Call Flow Examples October, 1999 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 INVITE F3 Proxy 2 -> B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: , From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 Proxy 1 -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 (100 Trying F5 Proxy 2 -> Proxy 1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy Johnston, et al. Informational [Page 67] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 (100 Trying F6 User B -> Proxy 2) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F7 B -> Proxy 2 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F8 Proxy 2 -> Proxy 1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F9 Proxy 1 Johnston, et al. Informational [Page 68] Internet Draft SIP Telephony Call Flow Examples October, 1999 -> A SIP/2.0 180 Ringing Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 480 Temporarily Unavailable F10 B -> Proxy 2 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP ss2.wcom.com:5060 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F11 Proxy 2 ->B ACK sip: UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss2.wcom.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 480 Temporarily Unavailable F12 Proxy 2 -> Proxy 1 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 69] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK F13 Proxy 1 -> Proxy 2 ACK sip: UserB@ss2.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 480 Temporarily Unavailable F14 Proxy1 -> A SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F15 A -> Proxy 1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy To: TheLittleGuy ;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 70] Internet Draft SIP Telephony Call Flow Examples October, 1999 4 SIP to Gateway Dialing In the following scenarios, User A (TheBigGuy sip:UserA@here.com) is a SIP phone or other SIP-enabled device. User B is reachable via the PSTN at global telephone number +1-972-555-2222. User A places a call to User B through a Proxy Server SS1 and a Network Gateway. In other scenarios, User A places calls to User C, who is served via a PBX (Private Branch Exchange) and is identified by a private extension 444-3333, or global number +1-918-555-3333. Note that User A uses his/her global telephone number +1-314-555-1111 in the From header in the INVITE messages. This then gives the Gateway the option of using this header to populate the calling party identification field in subsequent signaling (CgPN in ISUP). Left open is the issue of how the Gateway can determine the accuracy of the telephone number, necessary before passing it as a valid CgPN in the PSTN. Note that User A still uses his/her SIP URL in the Contact header. There is a major SIP issue in the call flows in this section and Section 6. In-band alerting (ringing tone, busy tone, recorded announcements, etc.) is present in the PSTN speech path after the receipt of the SS7 Address Complete Message (ACM) which maps to the SIP 180 Ringing response. In a SIP to SIP call, the media path is not established until the call is answered (200 OK sent). In order for the SIP caller User A to hear this alerting, it is necessary that an early media path be established to perform this. This is the purpose of the 183 Session Progress[5] responses used throughout this document in place of the 180 Ringinig. This document will be updated as this issue is further refined, with the possible inclusion of reliable responses[6] and/or additional SIP headers. 4.1 Success Scenarios In these scenarios, User A is a SIP phone or other SIP-enabled device. User A places a call to User B in the PSTN or User C on a PBX through a Proxy Server SS1 and a Gateway. Johnston, et al. Informational [Page 71] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.1.1 Successful SIP to ISUP PSTN call User A dials the globalized E.164 number +1-972-555-2222 to reach User B. Note that A might have only dialed the last 7 digits, or some other dialing plan. It is assumed that the SIP User Agent Client converts the digits into a global number and puts them into a SIP URL. User A could use either their SIP address (sip:UserA@here.com) or SIP telephone number (sip:+1-314-555-1111@ss1.wcom.com;user=phone) in the From header. In this example, the telephone number is included, and it is shown as being passed as calling party identification through the Network Gateway to User B (F5). Note that for this number to be passed into the SS7 network, it would have to be somehow verified for accuracy. In this scenario, User B answers the call then User A disconnects the call. Signaling between NGW1 and User B's telephone switch is SS7. Message Details INVITE F1 A->SS1 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 72] Internet Draft SIP Telephony Call Flow Examples October, 1999 (100 Trying F2 SS1 -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to NGW1. Client for A prepares to receive data on port 49170 from the network.*/ INVITE F3 SS1 -> NGW1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 GW -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone Johnston, et al. Informational [Page 73] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 IAM F5 GW -> User B IAM CdPN=972-555-2222,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required ACM F6 User B -> GW ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included 183 Session Progress F7 GW -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 74] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS1 proxies the OK to User A At this point the GW will start sendin an RTP path to the receive port on A encoding anything that is being received from B via the PSTN network (i.e. ringing) */ 183 Session Progress F8 SS1 ->User A SIP/2.0 183 Session Progress Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ANM F9 User B -> GW ANM 200 OK F10 GW -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Johnston, et al. Informational [Page 75] Internet Draft SIP Telephony Call Flow Examples October, 1999 Contact: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F11 SS1 ->User A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F12 A->SS1 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 76] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK F13 SS1 -> GW ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B(via the GW) */ /* User A Hangs Up with User B. */ BYE F14 A->SS1 BYE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 BYE F15 SS1 -> GW BYE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Johnston, et al. Informational [Page 77] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Length: 0 200 OK F16 GW -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F17 SS1->A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 REL F18 GW -> B REL CauseCode=16 Normal CodingStandard=CCITT RLC F19 B -> GW RLC Johnston, et al. Informational [Page 78] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.1.2 Successful SIP to ISDN PBX call User A is a SIP device while User C is connected via an Enterprise Gateway (GW1) to a PBX. The PBX connection is via a ISDN trunk group. User A dials User C's telephone number (918-555-3333) which is globalized and put into a SIP URL. The phone-context in the username portion of the Request-URI in message F3 is used to identify the context (customer, trunk group, or line) in which the private number 444-3333 is valid. Otherwise, this INVITE message could get forwarded and the context of the digits could become lost and the call unroutable. See section 1.1 for a discussion of phone-context. Proxy SS1 looks up the telephone number and locates the Enterprise Gateway that servers User C. User C is identified by its extension (444-3333) in the Request-URI sent to GW1. User A hears the ringing provided by the Gateway on the media path established after the 183 Session Progress response is received. Signaling between GW1 and PBX C is shown as ISDN. Message Details INVITE F1 A->SS1 INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 79] Internet Draft SIP Telephony Call Flow Examples October, 1999 (100 Trying F2 SS1 -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1 with the extension determined as 444-3333. Client for A prepares to receive data on port 49170 from the network.*/ INVITE F3 SS1 -> GW1 INVITE sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 GW -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 80] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 SETUP F5 GW -> User C Protocol discriminator=Q.931 Call reference: Flag=0, CR value=any valid value not in use Message type=SETUP Bearer capability: Information transfer capability=0 (Speech) or 16 (3.1 kHz audio) Channel identification=Preferred or exclusive B-channel Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) Called party number: Type of number and numbering plan ID=?? (private numbering plan) Digits=444-3333 CALL PROCeeding F6 User C -> GW Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F5 SETUP message Message type=CALL PROC Channel identification=Exclusive B-channel PROGress F7 User C -> GW Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F5 SETUP message Message type=PROG Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) 183 Session Progress F8 GW -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Johnston, et al. Informational [Page 81] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* The GW will establish an RTP path to the receive port on A encoding anything that is being received from C via the PSTN network (i.e. ringing) 183 Session Progress F9 SS1 ->User A SIP/2.0 183 Session Progress Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 CONNect F10 User C -> GW Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F5 SETUP message Message type=CONN CONNect ACK F11 GW -> User C Protocol discriminator=Q.931 Call reference: Flag=0, CR value=value in F5 SETUP message Message type=CONN ACK Johnston, et al. Informational [Page 82] Internet Draft SIP Telephony Call Flow Examples October, 1999 200 OK F12 GW -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: sip:444-3333,phone-context=p1234@gw1.wcom.com ;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F13 SS1 ->User A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: sip:444-3333,phone-context=p1234@gw1.wcom.com ;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F14 A->SS1 Johnston, et al. Informational [Page 83] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: ;user=phone From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 ACK F15 SS1 -> GW ACK sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B(via the GW) */ /* User A Hangs Up with User B. */ BYE F16 A->SS1 BYE sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: ;user=phone From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 Johnston, et al. Informational [Page 84] Internet Draft SIP Telephony Call Flow Examples October, 1999 BYE F17 SS1 -> GW BYE sip:444-3333,phone-context=p1234@gw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F18 GW -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F19 SS1->A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheOtherGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 DISConnect F20 GW -> User C Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F4 SETUP message Message type=DISC Cause=16 (Normal clearing) RELease F21 Johnston, et al. Informational [Page 85] Internet Draft SIP Telephony Call Flow Examples October, 1999 User C -> GW Protocol discriminator=Q.931 Call reference: Flag=0, CR value=value in F4 SETUP message Message type=REL RELease COMplete F22 GW -> User C Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F4 SETUP message Message type=REL COM Johnston, et al. Informational [Page 86] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.1.3 Successful SIP to ISUP PSTN call with overflow User A calls User B through SS1 working as a proxy server. SS1 tries an Enterprise Gateway GW1. GW1 is not available and responds with a 503 Service Unavailable (F4). The call is then routed to a Network Gateway NGW2. User B answers the call. The call is terminated when User A disconnects the call. NGW2 and User B's telephone switch use SS7 signaling. Message Details INVITE F1 A->SS1 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. SS1 receives a primary route NGW1 and a secondary route NGW2. NGW1 is tried first */ INVITE F2 SS1 -> NGW1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: Johnston, et al. Informational [Page 87] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F3 Proxy -> User A) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 503 Service Unavailable F4 NGW1-> SS1 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy GW2 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=123456789 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 /* SS1 now tries secondary route to NGW2 */ INVITE F6 SS1 -> NGW2 INVITE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 89] Internet Draft SIP Telephony Call Flow Examples October, 1999 IAM F7 NGW2 -> User B IAM CdPN=972-555-2222,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required ACM F8 User B -> NGW2 ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included 183 Session Progress F9 NGW2 -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 90] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* The GW will establish an RTP path to the receive port on A encoding anything that is being received from B via the PSTN network (i.e. ringing) */ 183 Session Progress F10 SS1 -> User A SIP/2.0 183 Session Progress Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ANM F11 User B -> NGW2 ANM 200 OK F12 NGW2 -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: sip:+1-972-555-2222@ngw2.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 Johnston, et al. Informational [Page 91] Internet Draft SIP Telephony Call Flow Examples October, 1999 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F13 SS1 -> User A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: sip:+1-972-555-2222@ngw2.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F14 A->SS1 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: ;user=phone From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 ACK F15 SS1 -> NGW2 Johnston, et al. Informational [Page 92] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B(via the GW) */ /* User A Hangs Up with User B. */ BYE F16 A->SS1 BYE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 Route: ;user=phone From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 BYE F17 SS1 -> NGW2 BYE sip:+1-972-555-2222@ngw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F18 NGW2 -> SS1 Johnston, et al. Informational [Page 93] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 200 OK F19 SS1-> User A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 2 BYE Content-Length: 0 REL F20 GW -> B REL CauseCode=16 Normal CodingStandard=CCITT RLC F21 B -> GW RLC Johnston, et al. Informational [Page 94] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.2 Failure Scenarios In these failure scenarios, the call does not complete. In most cases, however, a media stream is still setup. This is due to the fact that most failures in dialing to the PSTN result in in-band tones (busy, reorder tones) or announcements ("The number you have dialed has changed. The new number is..."). The 183 Session Progress[5] response containing SDP media information is used to setup this early media path so that the caller User A knows the final disposition of the call. The media stream is either terminated by the caller after the tone or announcement has been heard and understood, or by the Gateway after a timer expires. In other failure scenarios, a SS7 Release with Cause Code is mapped to a SIP response. In these scenarios, the early media path is not used, but the actual failure code is conveyed to the caller by the SIP User Agent Client. Johnston, et al. Informational [Page 95] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.2.1 Unsuccessful SIP to PSTN call: Treatment from PSTN User A calls User B in the PSTN through a proxy server SS1 and a Network Gateway NGW1. The call is rejected by the PSTN with an in- band treatment (tone or recording) played. User A hears the treatment and then issues a CANCEL (F9) to terminate the call. (A BYE is not sent since no final response was ever received by User A.) Message Details INVITE F1 A->SS1 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F2 SS1 -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 96] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to NGW1. Client for A prepares to receive data on port 49170 from the network.*/ INVITE F3 SS1 -> NGW1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 NGW1-> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 IAM F5 GW -> User B IAM CdPN=972-555-2222,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required Johnston, et al. Informational [Page 97] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACM F6 User B -> GW ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included 183 Session Progress F7 GW -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 183 Session Progress F8 SS1 ->User A SIP/2.0 183 Session Progress Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 Johnston, et al. Informational [Page 98] Internet Draft SIP Telephony Call Flow Examples October, 1999 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* User A listens to recorded announcement from the PSTN then hangs up */ CANCEL F9 A->SS1 CANCEL sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F10 SS1 -> GW CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 200 OK F11 GW -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone Johnston, et al. Informational [Page 99] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 200 OK F12 SS1->A SIP/2.0 200 OK Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 CANCEL Content-Length: 0 REL F13 GW -> B REL CauseCode=16 Normal CodingStandard=CCITT RLC F14 B -> GW RLC Johnston, et al. Informational [Page 100] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.2.2 Unsuccessful SIP to PSTN: REL w/Cause from PSTN User A calls PSTN User B through a Proxy Server SS1 and a Network Gateway NGW1. However, User A does not provide enough digits for the call to be completed. (In a real scenario, this call might have been rejected by SS1 based on incomplete address. However, especially on international calls, the number of digits in the number is not obvious, and this scenario may result.) The call is rejected by the PSTN with a SS7 Release message REL containing a specific Cause value. This cause value (28) is mapped by the Gateway to a SIP 484 Address Incomplete response which is proxied back to User A. Message Details INVITE F1 A->SS1 INVITE sip:+1-972-555-222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F2 SS1 -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 101] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to NGW1. Client for A prepares to receive data on port 49170 from the network. */ INVITE F3 SS1 -> NGW1 INVITE sip:+1-972-555-222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 NGW1-> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 IAM F5 GW -> User B IAM CdPN=972-555-2222,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National Johnston, et al. Informational [Page 102] Internet Draft SIP Telephony Call Flow Examples October, 1999 USI=Speech CPT=0 0 C=Normal CCI =Not Required REL F6 User B -> GW REL CauseValue=28 Address Incomplete CodingStandard=CCITT RLC F7 GW -> User B RLC /* Network Gateway maps CauseValue=28 to the SIP message 484 Address Incomplete */ 484 Address Incomplete F8 GW -> SS1 SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone; tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F9 SS1 -> GW ACK sip:+1-972-555-222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone; tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 484 Address Incomplete F10 SS1 -> User A Johnston, et al. Informational [Page 103] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 484 Address Incomplete Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone; tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F11 User A -> SS1 ACK sip:+1-972-555-222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone; tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 104] Internet Draft SIP Telephony Call Flow Examples October, 1999 4.2.3 Unsuccessful SIP to PSTN: ANM Timeout User A calls User B in the PSTN through a proxy server SS1 and a Gateway GW1. The call is released by the Gateway after its ISUP T9 timer expires due to no ANswer Message (ANM) being received. The Gateway sends a SS7 Release REL message to the PSTN and a 480 Temporarily Unavailable response to User A in the SIP network. Johnston, et al. Informational [Page 105] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details INVITE F1 A->SS1 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Authorization:Digest username="UserA", realm="MCI WorldCom SIP", nonce="ea9c8e88df84f1cec4341ae6cbe5a359", opaque="", uri="sip:ss1.wcom.com", response="dfe56131d1958046689cd83306477ecc" Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. Client for A prepares to receive data on port 49170 from the network.*/ (100 Trying F2 SS1 -> A) SIP/2.0 100 Trying Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 INVITE F3 SS1 -> GW1 Johnston, et al. Informational [Page 106] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 Record-Route: From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Contact: TheBigGuy Content-Type: application/sdp Content-Length: 132 v=0 o=UserA 2890844526 2890844526 IN IP4 here.com t=0 0 c=IN IP4 here.com m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 GW1 -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 IAM F5 GW -> User B IAM CdPN=972-555-2222,NPI=E.164,NOA=National CgPN=314-555-1111,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required ACM F6 User B -> GW ACM Charge Indicator=No Charge Called Party Status=no indication Johnston, et al. Informational [Page 107] Internet Draft SIP Telephony Call Flow Examples October, 1999 Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included 183 Session Progress F7 GW -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 183 Session Progress F8 SS1 -> User A SIP/2.0 183 Session Progress Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 108] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* After ISUP T9 Timer expires, Network Gateway sends REL to ISUP network and 480 to SIP network */ REL F9 GW -> User B REL CauseCode=16 Normal CodingStandard=CCITT RLC F10 User B -> GW RLC 480 Temporarily Unavailable F11 GW -> SS1 SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F12 SS1 -> GW ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 480 Temporarily Unavailable F13 SS1 -> User A SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Johnston, et al. Informational [Page 109] Internet Draft SIP Telephony Call Flow Examples October, 1999 Call-Id: 12345600@here.com CSeq: 1 INVITE Content-Length: 0 ACK F14 User A -> SS1 ACK sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP here.com:5060 From: TheBigGuy ;user=phone To: TheLittleGuy ;user=phone;tag=314159 Call-Id: 12345600@here.com CSeq: 1 ACK Content-Length: 0 Johnston, et al. Informational [Page 110] Internet Draft SIP Telephony Call Flow Examples October, 1999 5 Gateway to SIP Dialing 5.1 Success Scenarios In these scenarios, User A is placing calls from the PSTN to User B in a SIP network. User A's telephone switch signals to a Network Gateway (NGW1) using SS7. Since the called SIP User Agent does not send in-band signaling information, no early media path needs to be established on the IP side. As a result, the 183 Session Progress response is not used. However, NGW1 will establish a one way speech path prior to call completion, and generate ringing for the PSTN caller. Any tones or recordings are generated by NGW1 and played in this speech path. When the call completes successfully, NGW1 bridges the PSTN speech path with the IP media path. Johnston, et al. Informational [Page 111] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.1.1 Successful PSTN to SIP call In this scenario, User A from the PSTN calls User B through a Network Gateway NGW1 and Proxy Server SS1. When User B answers the call the media path is setup end-to-end. The call terminates when User A hangs up the call, with User A's telephone switch sending a SS7 Release message which is mapped to a BYE by NGW1. Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ Johnston, et al. Informational [Page 112] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE F3 SS1 ->User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 User B -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F5 User B -> SS1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com>;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F6 SS1 Johnston, et al. Informational [Page 113] Internet Draft SIP Telephony Call Flow Examples October, 1999 -> NGW1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACM F7 NGW1-> User A ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included 200 OK F8 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com Contact: TheLittleGuy CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F9 SS1 -> NGW1 SIP/2.0 200 OK Johnston, et al. Informational [Page 114] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F10 GW1 -> SS1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACK F11 SS1 -> User B ACK sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 ANM F12 User B -> NGW1 ANM Johnston, et al. Informational [Page 115] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* RTP streams are established between A and B (via the GW) */ /* User A Hangs Up with User B. */ REL F13 User A -> NGW1 REL CauseCode=16 Normal CodingStandard=CCITT RLC F14 NGW1-> User A RLC BYE F15 NGW1-> SS1 BYE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 BYE F16 SS1 -> User B BYE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F17 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Johnston, et al. Informational [Page 116] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F18 SS1 -> NGW1 SIP/2.0 200 OK Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 Johnston, et al. Informational [Page 117] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.1.2 Successful PSTN to SIP call, Fast Answer This "fast answer" scenario is similar to 5.1.1 except that User B immediately accepts the call, sending a 200 OK (F5) without sending a 180 Ringing response. The Gateway then sends an Answer Message (ANM) without sending an Address Complete Message (ACM). Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 GW -> SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to User B. User B prepares to receive data on port 3456 from User A.*/ Johnston, et al. Informational [Page 118] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE F3 SS1 -> User B INVITE UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 SS1 -> GW1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 200 OK F5 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 150 Johnston, et al. Informational [Page 119] Internet Draft SIP Telephony Call Flow Examples October, 1999 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F6 SS1 -> GW1 SIP/2.0 200 OK Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: TheLittleGuy Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F7 GW1 ->SS1 ACK UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACK F8 SS1 ->User B ACK UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Johnston, et al. Informational [Page 120] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 ANM F9 User B ->GW ANM /* RTP streams are established between A and B (via the GW) */ /* User A Hangs Up with User B. */ REL F10 User A ->GW REL CauseCode=16 Normal CodingStandard=CCITT RLC F11 GW -> User A RLC BYE F12 GW -> SS1 BYE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 BYE F13 SS1 -> User B BYE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Johnston, et al. Informational [Page 121] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F14 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F15 SS1 ->GW SIP/2.0 200 OK Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 Johnston, et al. Informational [Page 122] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.1.3 Successful PBX to SIP call In this scenario, User A calls from PBX A to User B through GW1 and SS1 working as a proxy server. Signaling between PBX A and GW1 is Feature Group B (FGB) circuit associated signaling (in-band mult- frequency outpulsing). After the receipt of the 180 Ringing from User B, GW1 generates ringing tone for User A. User B answers the call by sending a 200 OK. The call terminates when User A hangs up, causing GW1 to send a BYE. Message Details MF Digits F1 PBX A ->GW1 KP 1 972 555 2222 ST INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Contact: PBX_A ;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1.*/ Johnston, et al. Informational [Page 123] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE F3 SS1 ->User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Contact: PBX_A ;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 SS1 -> GW) SIP/2.0 100 Trying Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F5 User B -> SS1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone ;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F6 SS1 Johnston, et al. Informational [Page 124] Internet Draft SIP Telephony Call Flow Examples October, 1999 -> GW1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Content-Length: 0 /* One way Voice path is established between GW and the PBX for ringing. */ 200 OK F7 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com Contact: TheLittleGuy CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F8 SS1 -> GW1 SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 1 INVITE Contact: TheLittleGuy Johnston, et al. Informational [Page 125] Internet Draft SIP Telephony Call Flow Examples October, 1999 Content-Type: application/sdp Content-Length: 134 v=0 o=UserB 2890844527 2890844527 IN IP4 there.com t=0 0 c=IN IP4 110.111.112.113 m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ACK F9 GW1 ->SS1 ACK sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP gw1.wcom.com:5060 Route: From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACK F10 SS1 ->User B ACK sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between A and B (via the GW) */ /* User A Hangs Up with User B. */ BYE F11 GW -> SS1 BYE sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP gw1.wcom.com:5060 Route: Johnston, et al. Informational [Page 126] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 2 BYE Content-Length: 0 BYE F12 SS1 -> User B BYE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F13 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 2 BYE Content-Length: 0 200 OK F14 SS1 ->GW SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A ;user=phone To: sip:+1-972-555-2222@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@gw1.wcom.com CSeq: 2 BYE Content-Length: 0 Johnston, et al. Informational [Page 127] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2 Failure Scenarios 5.2.1 Unsuccessful PSTN to SIP REL, SIP error mapped to REL User A attempts to call a SIP user through SS1 working as a proxy server. SS1 is unable to find any routing for the number. The call is rejected by SS1 with a REL message containing a specific Cause value mapped by the gateway based on the SIP error. Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-9999,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 A->SS1 INVITE sip:+1-972-555-9999@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 128] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ 604 Does Not Exist Anwhere F3 SS1 -> GW SIP/2.0 604 Does Not Exist Anywhere Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACK F4 GW1 ->SS1 ACK sip:+1-972-555-9999@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-9999@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 REL F5 GW -> User A REL CauseCode=1 CodingStandard=CCITT RLC F6 User A -> GW RLC Johnston, et al. Informational [Page 129] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2.2 Unsuccessful PSTN to SIP REL, SIP busy mapped to REL In this scenario, User A calls User B through a Gateway GW1 and SS1 working as a proxy server. The call is routed to User B via the gateway. The call is rejected by the User B who sends a 600 Busy Everywhere response. The Gateway sends a REL message containing a specific Cause value mapped by the gateway based on the SIP error. Since no interworking is indicated in the IAM (F1), the busy tone is generated locally by User A's telephone switch. In scenario 5.2.3, the busy signal is generated by the Gateway since interworking is indicated. Johnston, et al. Informational [Page 130] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ INVITE F3 SS1 ->User B INVITE UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 131] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 SS1 -> GW1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 600 Busy Everywhere F5 User B -> SS1 SIP/2.0 600 Busy Everywhere Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACK F6 SS1 ->User B ACK UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Johnston, et al. Informational [Page 132] Internet Draft SIP Telephony Call Flow Examples October, 1999 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 600 Busy Everywhere F7 SS1 -> GW1 SIP/2.0 600 Busy Everywhere Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACK F8 GW1 ->SS1 ACK UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 REL F9 User A ->GW REL CauseCode=17 Busy CodingStandard=CCITT RLC F10 GW -> User A RLC Johnston, et al. Informational [Page 133] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2.3 Unsuccessful PSTN->SIP, SIP error interworking to tones In this scenario, User A calls User B through SS1 working as a proxy server. The call is routed to User B via the gateway. The call is rejected by the User B client. GW1 plays busy tone, and releases call after timeout. GW1 plays the busy tone since the IAM (F1) indicates the interworking is present. In scenario 5.2.2, with no interworking, the busy indication is carried in the REL Cause value and is generated locally instead. Johnston, et al. Informational [Page 134] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required Interworking=encountered INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ INVITE F3 SS1 ->User B INVITE UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 135] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 User B -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 600 Busy Everywhere F5 User B -> SS1 SIP/2.0 600 Busy Everywhere Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACK F6 SS1 ->User B ACK UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 136] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 600 Busy Everywhere F7 SS1 -> GW1 SIP/2.0 600 Busy Everywhere Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACK F8 GW1 ->SS1 ACK sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACM F9 User B ->GW ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included /* One way speech path established between GW and User A. */ Johnston, et al. Informational [Page 137] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* Call Released after NGW treatment timer expires. */ REL F10 User A ->GW REL CauseCode=17 CodingStandard=CCITT RLC F11 GW -> User A RLC Johnston, et al. Informational [Page 138] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2.4 Unsuccessful PSTN->SIP, ACM timeout User A calls User B through SS1 working as a proxy server. SS1 re- sends the INVITE after the expiration of SIP timer T1. User B never responds with 180 Ringing _ it is reachable but unresponsive. After the expiration of ISUP T7 timer, User A's network disconnects the call by sending a Release message REL. The Gateway maps this to a CANCEL which is re-sent by SS1 after SIP T1 timer expires. Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 139] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ INVITE F3 SS1 ->User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com c t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 100 Trying F4 SS1 -> GW SIP/2.0 100 Trying Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 INVITE F5 GW -> SS1 Same as Message F3 INVITE F6 Johnston, et al. Informational [Page 140] Internet Draft SIP Telephony Call Flow Examples October, 1999 SS1 ->User B Same as Message F3 INVITE F7 SS1 ->User B Same as Message F3 INVITE F8 SS1 ->User B Same as Message F3 INVITE F9 SS1 ->User B Same as Message F3 /* ISUP Timer T7 expires in User A's access network. */ REL F10 User A ->GW REL CauseCode=16 Normal CodingStandard=CCITT RLC F11 GW -> User A RLC CANCEL F12 GW -> SS1 CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Johnston, et al. Informational [Page 141] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F13 SS1 -> User B CANCEL sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F14 GW -> SS1 Same as Message F13 CANCEL F15 SS1 ->User B Same as Message F13 CANCEL F16 SS1 ->User B Same as Message F13 CANCEL F17 SS1 ->User B Same as Message F13 CANCEL F18 SS1 Johnston, et al. Informational [Page 142] Internet Draft SIP Telephony Call Flow Examples October, 1999 ->User B Same as Message F13 Johnston, et al. Informational [Page 143] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2.5 Unsuccessful PSTN->SIP, ACM timeout, stateless SPS In this scenario, User A calls User B through SS1 working as a stateless proxy server. Since SS1 is stateless, GW1 re-sends the INVITE and CANCEL messages after the expiration of SIP timer T1. User B does not respond with 180 Ringing. User A's network disconnects the call with a release REL. Johnston, et al. Informational [Page 144] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 GW -> SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ INVITE F3 SS1-> User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: Johnston, et al. Informational [Page 145] Internet Draft SIP Telephony Call Flow Examples October, 1999 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 INVITE F4 GW -> SS1 Same as Message F2 INVITE F5 SS1 -> User B Same as Message F3 INVITE F6 GW -> SS1 Same as Message F2 INVITE F7 SS1 -> User B Same as Message F3 INVITE F8 GW -> SS1 Same as Message F2 INVITE F9 SS1 -> User B Same as Message F3 Johnston, et al. Informational [Page 146] Internet Draft SIP Telephony Call Flow Examples October, 1999 INVITE F10 GW -> SS1 Same as Message F2 INVITE F11 SS1 -> User B Same as Message F3 INVITE F12 GW -> SS1 Same as Message F2 INVITE F13 SS1 -> User B Same as Message F3 /* ISUP T7 Timer expires in User A's access network. */ REL F14 User A ->GW REL CauseCode=16 Normal CodingStandard=CCITT RLC F15 GW -> User A RLC CANCEL F16 GW -> SS1 CANCEL sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com Johnston, et al. Informational [Page 147] Internet Draft SIP Telephony Call Flow Examples October, 1999 CSeq: 1 CANCEL Content-Length: 0 CANCEL F17 SS1 -> User B CANCEL sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F18 GW -> SS1 Same as Message F16 CANCEL F19 SS1 -> User B Same as Message F17 CANCEL F20 GW -> SS1 Same as Message F16 CANCEL F21 SS1 -> User B Same as Message F17 CANCEL F22 GW -> SS1 Same as Message F16 CANCEL F23 SS1 -> User B Same as Message F17 CANCEL F24 GW -> SS1 Same as Message F16 CANCEL F25 SS1 -> User B Same as Message F17 CANCEL F26 Johnston, et al. Informational [Page 148] Internet Draft SIP Telephony Call Flow Examples October, 1999 GW -> SS1 Same as Message F16 CANCEL F27 SS1 -> User B Same as Message F17 Johnston, et al. Informational [Page 149] Internet Draft SIP Telephony Call Flow Examples October, 1999 5.2.6 Unsuccessful PSTN->SIP, ANM timeout In this scenario, User A calls User B through SS1 working as a proxy server. User B does not respond with 200 OK. User A disconnects the call with a Release message REL which is mapped by GW1 to a CANCEL. Note that if User B had sent a 200 OK response after the REL, GW1 would have sent an ACK then a BYE to properly terminate the call. Johnston, et al. Informational [Page 150] Internet Draft SIP Telephony Call Flow Examples October, 1999 Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 A->SS1 INVITE sip:+1-972-555-2222@ngw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS1 uses a location manager function to determine where B is actually located. Based upon location analysis the call is forwarded to GW1. GW1 prepares to receive data on port 3456 from User A.*/ INVITE F3 SS1 ->User B INVITE sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Johnston, et al. Informational [Page 151] Internet Draft SIP Telephony Call Flow Examples October, 1999 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 150 v=0 o=GATEWAY1 2890844527 2890844527 IN IP4 gatewayone.wcom.com t=0 0 c=IN IP4 gatewayone.wcom.com m=audio 3456 RTP/AVP 0 a=rtpmap:0 PCMU/8000 (100 Trying F4 User B -> SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F5 User B -> SS1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 180 Ringing F6 SS1 -> GW1 SIP/2.0 180 Ringing Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Johnston, et al. Informational [Page 152] Internet Draft SIP Telephony Call Flow Examples October, 1999 To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 INVITE Content-Length: 0 ACM F7 GW -> User A ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included /* ISUP Timer T9 expires in User A's access network. */ REL F8 User A -> GW REL CauseCode=16 Normal CodingStandard=CCITT RLC F9 GW -> User A RLC CANCEL F10 GW -> SS1 CANCEL sip:UserB@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 CANCEL F11 SS1 -> User B Johnston, et al. Informational [Page 153] Internet Draft SIP Telephony Call Flow Examples October, 1999 CANCEL sip:UserB@there.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 200 OK F12 User B -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 200 OK F13 SS1 -> GW SIP/2.0 200 OK Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-972-555-2222@ngw1.wcom.com;user=phone;tag=314159 Call-Id: 12345602@ngw1.wcom.com CSeq: 1 CANCEL Content-Length: 0 Johnston, et al. Informational [Page 154] Internet Draft SIP Telephony Call Flow Examples October, 1999 6 Gateway to Gateway Dialing via SIP Network In these scenarios, both the caller and the called party are in the telephone network, either normal PSTN subscribers or PBX extensions. The calls route through two Gateways and at least one SIP Proxy Server. The Proxy Server performs the authentication and location of the Gateways. Note that the proposed INFO method[8] is not currently included in this document. It is anticipated that a future version will include an example of this. Again it is noted that the intent of this call flows document is not to provide a detailed parameter level mapping of SIP to PSTN protocols. For information on SIP to ISUP mapping, the reader is referred to other references[9]. 6.1 Success Scenarios In these scenarios, the call is successfully completed between the two Gateways allowing the PSTN or PBX users to communicate. The 183 Session Progress response is used to establish a media path between the two Gateways, allowing in-band alerting to pass from the called party telephone switch to the caller. Johnston, et al. Informational [Page 155] Internet Draft SIP Telephony Call Flow Examples October, 1999 6.1.1 Successful ISUP PSTN to ISUP PSTN call In this scenario, User A in the PSTN calls User C who is served as an extension on a PBX. User A's telephone switch signals via SS7 to the Network Gateway NGW1, while User C's PBX signals via SS7 with the Enterprise Gateway GW2. The CdPN and CgPN are mapped into SIP URLs and placed in the To and From headers. SS1 looks up the dialed digits in the Request-URI and maps the digits to the PBX extension of User C served by GW2. The INVITE is then forwarded to GW2 for call completion. An early media path is established end-to-end so that User A can hear the ringing tone generated by PBX C. User C answers the call and the media path is cut through in both directions. User B hangs up terminating the call. Message Details IAM F1 User A -> GW IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=972-555-2222,NPI=E.164,NOA=National USI=Speech CPT=0 0 C=Normal CCI =Not Required INVITE F2 GW1 -> SS1 INVITE sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 Johnston, et al. Informational [Page 156] Internet Draft SIP Telephony Call Flow Examples October, 1999 /* SS uses a location manager function to determine where B is actually located. Response is returned listing onnet and offnet routes. */ INVITE F3 SS1 -> GW2 INVITE sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 o=GW1 2890844526 2890844526 IN IP4 gw1.wcom.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 IAM F4 GW2 -> User C IAM CgPN=314-555-1111,NPI=E.164,NOA=National CdPN=444-3333,NPI=Private,NOA=Subscriber USI=Speech CPT=0 0 C=Normal CCI =Not Required ACM F5 User C -> GW2 ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber Johnston, et al. Informational [Page 157] Internet Draft SIP Telephony Call Flow Examples October, 1999 End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included /* Based on PROGress message, GW3 returns a 183 response with SDP allowing in-band call progress indications to be sent to the originator. */ 183 Session Progress F6 GW2 -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw2.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 183 Session Progress F7 SS1 -> GW1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw2.wcom.com t=0 0 c=IN IP4 100.101.102.104 Johnston, et al. Informational [Page 158] Internet Draft SIP Telephony Call Flow Examples October, 1999 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* GW1 receives packets from GW2 with encoded ringback, tones or other audio. GW1 decodes this and places it on the originating trunk. */ ACM F8 GW1 ->User A ACM Charge Indicator=No Charge Called Party Status=no indication Called Party's Category=ordinary subscriber End To End Method=none available Interworking=encountered End to End Information=none available ISUP Indicator=not used all the way ISDN Access Terminating access non ISDN Echo Control=not included /* User B answers */ ANM F9 User C -> GW2 ANM 200 OK F10 GW2 -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 Johnston, et al. Informational [Page 159] Internet Draft SIP Telephony Call Flow Examples October, 1999 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F11 SS1 -> GW1 SIP/2.0 200 OK Via: SIP/2.0/UDP ngw1.wcom.com:5060 Record-Route: From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@ngw1.wcom.com CSeq: 1 INVITE Contact: sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ANM F12 GW1 -> User A ANM ACK F13 GW1 -> SS1 ACK sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ngw1.wcom.com:5060 Route: ;user=phone From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@gw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACK F14 SS1 -> GW3 Johnston, et al. Informational [Page 160] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP ngw1.wcom.com:5060 From: sip:+1-314-555-1111@ngw1.wcom.com;user=phone To: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 Call-Id: 12345600@ngw1.wcom.com CSeq: 1 ACK Content-Length: 0 /* RTP streams are established between GW1 and GW2. */ /* User B Hangs Up with User A. */ REL F15 User C-> GW2 REL CauseCode=16 Normal CodingStandard=CCITT BYE F16 GW3 -> SS1 BYE sip:+1-314-555-1111@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP gw2.wcom.com:5060 Route: ;user=phone From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 4 BYE Content-Length: 0 RLC F17 GW2 ->User C RLC BYE F18 SS1 -> GW1 BYE sip:+1-314-555-1111@gw1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Johnston, et al. Informational [Page 161] Internet Draft SIP Telephony Call Flow Examples October, 1999 Via: SIP/2.0/UDP gw2.wcom.com:5060 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 4 BYE Content-Length: 0 200 OK F19 GW1 -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw2.wcom.com:5060 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 4 BYE Content-Length: 0 200 OK F20 SS11 ->GW3 SIP/2.0 200 OK Via: SIP/2.0/UDP gw2.wcom.com:5060 From: sip:+1-918-555-3333@ss1.wcom.com;user=phone;tag=314159 To: sip:+1-314-555-1111@ngw1.wcom.com;user=phone Call-Id: 12345600@ngw1.wcom.com CSeq: 4 BYE Content-Length: 0 REL F21 User C-> GW2 REL CauseCode=16 Normal CodingStandard=CCITT RLC F22 GW2 ->User C RLC Johnston, et al. Informational [Page 162] Internet Draft SIP Telephony Call Flow Examples October, 1999 6.1.2 Successful FGB PBX to ISDN PBX call with overflow PBX User A calls PBX User C via GW1 using SS1 as a Proxy Server. During the attempt to reach User C via GW2, an error is encountered _ SS1 receives a 503 Service Unavailable (F4) response to the forwarded INVITE. This could be due to all circuits being busy, or some other outage at GW2. SS1 recognizes the error and uses an alternative route via GW3 to terminate the call. From there, the call proceeds normally with User C answering the call. The call is terminated when User C hangs up. Message Details PBX A ->GW1 Seizure GW1 -> PBX A Wink MF Digits F1 PBX A ->GW1 KP 444 3333 ST INVITE F2 GW1 -> SS1 INVITE sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Contact: PBX_A Content-Type: application/sdp Content-Length: 136 v=0 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com t=0 0 Johnston, et al. Informational [Page 163] Internet Draft SIP Telephony Call Flow Examples October, 1999 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* SS uses a location manager function to determine where B is actually located. Response is returned listing onnet and offnet routes. */ INVITE F3 SS1 -> GW2 INVITE sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Contact: PBX_A Content-Type: application/sdp Content-Length: 136 v=0 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 503 Service Unavailable F4 GW2 -> SS1 SIP/2.0 503 Service Unavailable Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=314159 Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Content-Length: 0 Johnston, et al. Informational [Page 164] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK F5 SS1 -> GW2 ACK sip:444-3333,phone-context=p1234@gw2.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=314159 Call-Id: 12345600@gw1.wcom.com CSeq: 1 ACK Content-Length: 0 INVITE F6 SS1 -> GW3 INVITE sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Contact: PBX_A Content-Type: application/sdp Content-Length: 136 v=0 o=PBX_A 2890844526 2890844526 IN IP4 gw1.wcom.com t=0 0 c=IN IP4 100.101.102.103 m=audio 49170 RTP/AVP 0 a=rtpmap:0 PCMU/8000 SETUP F7 GW3-> PBX C Protocol discriminator=Q.931 Call reference: Flag=0, CR value=any valid value not in use Message type=SETUP Bearer capability: Information transfer capability=0 (Speech) or 16 (3.1 kHz audio) Channel identification=Preferred or exclusive B-channel Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) Called party number: Johnston, et al. Informational [Page 165] Internet Draft SIP Telephony Call Flow Examples October, 1999 Type of number and numbering plan ID=33 (National number in ISDN numbering plan) Digits=918-555-3333 (100 Trying F8 GW3 ->SS1) SIP/2.0 100 Trying Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com;user=phone Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Content-Length: 0 CALL PROCeeding F9 PBX C -> GW3 Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F9 SETUP message Message type=CALL PROC Channel identification=Exclusive B-channel PROGress F10 PBX C -> GW3 Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F9 SETUP message Message type=PROG Progress indicator=1 (Call is not end-to-end ISDN;further call progress information may be available inband) /* Based on PROGress message, GW3 returns a 183 response with SDP allowing in-band call progress indications to be sent to the originator. */ 183 Session Progress F11 GW3 -> SS1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Johnston, et al. Informational [Page 166] Internet Draft SIP Telephony Call Flow Examples October, 1999 Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 183 Session Progress F12 SS1 -> GW1 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 /* GW1 receives packets from GW3 with encoded ringback, tones or other audio. GW1 decodes this and places it on the originating trunk. */ CONNect F13 PBX C -> GW3 Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F9 SETUP message Message type=CONN 200 OK F14 GW3 -> SS1 Johnston, et al. Informational [Page 167] Internet Draft SIP Telephony Call Flow Examples October, 1999 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-918-555-3333@gw3.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 200 OK F15 SS1 -> GW1 SIP/2.0 200 OK Via: SIP/2.0/UDP gw1.wcom.com:5060 Record-Route: From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Call-Id: 12345600@gw1.wcom.com CSeq: 1 INVITE Contact: sip:+1-918-555-3333@gw3.wcom.com;user=phone Content-Type: application/sdp Content-Length: 134 v=0 o=PBX_B 987654321 987654321 IN IP4 gw3.wcom.com t=0 0 c=IN IP4 100.101.102.104 m=audio 14918 RTP/AVP 0 a=rtpmap:0 PCMU/8000 GW1 -> PBX A Seizure ACK F16 GW1 -> SS1 Johnston, et al. Informational [Page 168] Internet Draft SIP Telephony Call Flow Examples October, 1999 ACK sip:+1-918-555-3333@ss1.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP gw1.wcom.com:5060 Route: From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Call-Id: 12345600@gw1.wcom.com CSeq: 1 ACK Content-Length: 0 ACK F17 SS1 -> GW3 ACK sip:+1-918-555-3333@gw3.wcom.com;user=phone SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw1.wcom.com:5060 From: PBX_A To: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 Call-Id: 12345600@gw1.wcom.com CSeq: 1 ACK Content-Length: 0 CONNect ACK F18 GW3-> PBX C Protocol discriminator=Q.931 Call reference: Flag=0, CR value=value in F9 SETUP message Message type=CONN ACK /* RTP streams are established between GW1 and GW3. */ /* User B Hangs Up with User A. */ DISConnect F19 PBX C -> GW3 Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F9 SETUP message Message type=DISC Cause=16 (Normal clearing) BYE F20 Johnston, et al. Informational [Page 169] Internet Draft SIP Telephony Call Flow Examples October, 1999 GW3 -> SS1 BYE sip:IdentifierString@ss1.wcom.com SIP/2.0 Via: SIP/2.0/UDP gw3.wcom.com:5060 Route: From: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 To: PBX_A Call-Id: 12345600@gw1.wcom.com CSeq: 1 BYE Content-Length: 0 BYE F21 SS1 -> GW1 BYE sip:IdentifierString@gw1.wcom.com SIP/2.0 Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw3.wcom.com:5060 From: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 To: PBX_A Call-Id: 12345600@gw1.wcom.com CSeq: 1 BYE Content-Length: 0 GW1 -> PBX A Seizure removal RELease F22 GW3-> PBX C Protocol discriminator=Q.931 Call reference: Flag=0, CR value=value in F9 SETUP message Message type=REL 200 OK F23 GW1 -> SS1 SIP/2.0 200 OK Via: SIP/2.0/UDP ss1.wcom.com:5060 Via: SIP/2.0/UDP gw3.wcom.com:5060 From: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 To: PBX_A Call-Id: 12345600@gw1.wcom.com CSeq: 1 BYE Content-Length: 0 Johnston, et al. Informational [Page 170] Internet Draft SIP Telephony Call Flow Examples October, 1999 200 OK F24 SS11 ->GW3 SIP/2.0 200 OK Via: SIP/2.0/UDP gw3.wcom.com:5060 From: sip:444-3333,phone-context=p1234@ss1.wcom.com ;user=phone;tag=123456789 To: PBX_A Call-Id: 12345600@gw1.wcom.com CSeq: 1 BYE Content-Length: 0 RELease COMplete F25 PBX C -> GW3 Protocol discriminator=Q.931 Call reference: Flag=1, CR value=value in F9 SETUP message Message type=REL COM PBX A ->GW1 Seizure removal Johnston, et al. Informational [Page 171] Internet Draft SIP Telephony Call Flow Examples October, 1999 7 Acknowledgements The authors wish to thank the following individuals for their assistance and review of this call flows document: Dean Willis, Henry Sinnreich, David Devanatham, Joe Pizzimenti, Matt Cannon, John Hearty, the whole MCI WorldCom IPOP Design team, and from Nortel Networks: Scott Orton, Greg Osterhout, Pat Sollee, Doug Weisenberg, Danny Mistry, Steve McKinnon, and Denise Ingram. 8 References [1] S. Bradner, "The Internet Standards Process -- Revision 3", BCP 9, RFC 2026, October 1996. [2] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP: Session Initiation Protocol", RFC 2543, March 1999. [3] R. Sparks, C. Cunningham, A. Johnston, S. Donovan, D. Willis, and K. Summers, "SIP Telephony Service Examples with Call Flows", Internet Draft, Internet Engineering Task Force, October 1999. Work in progress. [4] S. Kent, R. Atkinson, "Security Architecture for the Internet Protocol", RFC 2401, November 1998. [5] S. Donovan, J. Hearty, M. Cannon, H. Schulzrinne, and J. Rosenberg, "SIP 183 Session Progress Message", Internet Draft, Internet Engineering Task Force, June 1999. Work in progress. [6] J. Rosenberg, and H. Schulzrinne, "Reliability of Provisional Responses in SIP", Internet Draft, Internet Engineering Task Force, May 20, 1999, Work in progress. [7] A. Vaha-Sipila, "URLs for Telephone Calls", Internet Draft, Internet Engineering Task Force, September 1999, Work in progress [8] S. Donovan, M. Cannon, "The SIP INFO Method", Internet Draft, Internet Engineering Task Force, June 1999. Work in progress. [9] G. Camarillo, "Best Current Practice for ISUP to SIP Mapping", Internet Draft, Internet Engineering Task Force, August 1999, Work in progress. Johnston, et al. Informational [Page 172] Internet Draft SIP Telephony Call Flow Examples October, 1999 Author's Addresses Alan Johnston MCI WorldCom 100 S 4th Street Phone: +1-314-342-7360 St. Louis, MO 63104 Email: alan.johnston@wcom.com Steve Donovan MCI WorldCom 901 International Parkway Phone: +1-972-729-1621 Richardson, TX 65081 Email: steven.r.donovan@wcom.com Robert Sparks MCI WorldCom 2400 N Glenville Drive Phone: +1-972-729-5241 Richardson, TX 75082 Email: Robert.Sparks@wcom.com Chris Cunningham MCI WorldCom 400 International Parkway Phone: +1-972-729-3110 Richardson, TX 75081 Email: Chris.Cunningham@wcom.com Kevin Summers MCI WorldCom 2400 N Glenvile Drive Phone: +1-972-729-7976 Richardson, TX 75082 Email: Kevin.Summers@wcom.com Copyright Notice "Copyright (C) The Internet Society 1999. All Rights Reserved. This document and translations of it may be copied and furnished to others, and derivative works that comment on or otherwise explain it or assist in its implementation may be prepared, copied, published and distributed, in whole or in part, without restriction of any kind, provided that the above copyright notice and this paragraph are included on all such copies and derivative works. However, this document itself may not be modified in any way, such as by removing the copyright notice or references to the Internet Society or other Internet organizations, except as needed for the purpose of developing Internet standards in which case the procedures for copyrights defined in the Internet Standards process must be followed, or as required to translate it into languages other than English. The limited permissions granted above are perpetual and will not be revoked by the Internet Society or its successors or assigns. Johnston, et al. Informational [Page 173] Internet Draft SIP Telephony Call Flow Examples October, 1999 This document and the information contained herein is provided on an "AS IS" basis and THE INTERNET SOCIETY AND THE INTERNET ENGINEERING TASK FORCE DISCLAIMS ALL WARRANTIES, EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE. Johnston, et al. Informational [Page 174]