Yep, no problem. Here it is. Same endpoints. The only thing that changed is the call direction. Received Message : SIP/2.0 200 OK Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bK5Qtv3B04eS46g From: "Test 2" <sip:xxxxxxxx75@A.B.C.D>;tag=yy2XQKXSy7gtN To: <sip:xxxxx02@I.J.K.L>;tag=93BEF648-1A8B Date: Tue, 24 May 2011 16:37:23 GMT Call-ID: b7967af6-00c4-122f-19ba-000c29c18d38 CSeq: 12790979 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Remote-Party-ID: <sip:xxxxx02@I.J.K.L>;party=called;screen=no;privacy=off Contact: <sip:xxxxx02@I.J.K.L:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 334 v=0 o=CiscoSystemsSIP-GW-UserAgent 6942 3511 IN IP4 I.J.K.L s=SIP Call c=IN IP4 I.J.K.L t=0 0 m=audio 17046 RTP/AVP 18 19 101 100 c=IN IP4 I.J.K.L a=rtpmap:18 G729/8000 a=rtpmap:19 CN/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 Sent Message : SIP/2.0 200 OK Via: SIP/2.0/UDP E.F.G.H:5060;branch=z9hG4bK9E31DBF From: "Test 2" <sip:xxxxxxxx75@A.B.C.D>;tag=65BCD9B0-115E To: <sip:xxxxx02@A.B.C.D>;tag=XN94Nrcp1yt7S Call-ID: A4A3C9E0-855A11E0-8237C79A-B2A2FA42@E.F.G.H CSeq: 102 INVITE Contact: <sip:xxxxx02@A.B.C.D:5060;transport=udp> User-Agent: GCI Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 333 Remote-Party-ID: "xxxxx02" <sip:xxxxx02@A.B.C.D>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 3912792884 3912792885 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 27728 RTP/AVP 18 100 19 101 c=IN IP4 A.B.C.D a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of David Ponzone Sent: Monday, May 23, 2011 09:42 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] rtpmap line missing on answer Sean, can you show us the same packets, but for an incoming call (that works, if I understood you correctly) ? David Ponzone Direction Technique email: david.ponzone@ipeva.fr<mailto:david.ponzone@ipeva.fr> tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr<BLOCKED::http://www.ipeva.fr/> - www.ipeva-studio.com<BLOCKED::http://www.ipeva-studio.com/> Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 24/05/2011 à 01:53, Sean Eichhorn a écrit : Yeah, they’re negotiating the same codec on both sides. The problem is the way Cisco handles the negotiation. Here’s an example of what I see in Freeswitch: Received from CALLED : SIP/2.0 200 OK Via: SIP/2.0/UDP A.B.C.D;rport;branch=z9hG4bKXZZ18p8tD03cS From: "pending" <sip:xxxxxxxx02@A.B.C.D>;tag=NH1HrNe744HSB To: <sip:sean@E.F.G.H:5060>;tag=622555CC-408 Date: Mon, 23 May 2011 23:50:56 GMT Call-ID: 69258f8c-0038-122f-19ba-000c29c18d38 CSeq: 12760849 INVITE Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER Allow-Events: telephone-event Contact: <sip:xxxxxxxx75@E.F.G.H:5060> Supported: replaces Supported: sdp-anat Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp Content-Disposition: session;handling=required Content-Length: 311 v=0 o=CiscoSystemsSIP-GW-UserAgent 4080 7999 IN IP4 E.F.G.H s=SIP Call c=IN IP4 E.F.G.H t=0 0 m=audio 17122 RTP/AVP 0 19 101 100 c=IN IP4 E.F.G.H a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:100 X-NSE/8000 a=fmtp:100 192-194 Sent to CALLING : SIP/2.0 200 OK Via: SIP/2.0/UDP I.J.K.L:5060;branch=z9hG4bK4C2B6C64 From: <sip:xxxxxxxx02@ I.J.K.L >;tag=9020153C-21B3 To: <sip:xxxxxxxx75@A.B.C.D>;tag=m87rptX37UU6F Call-ID: AFADEF2E-84CC11E0-B1D4F23A-27804614@ I.J.K.L CSeq: 101 INVITE Contact: <sip:xxxxxxxx75@ A.B.C.D:5060;transport=udp> User-Agent: GCI Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Min-SE: 1800 Content-Type: application/sdp Content-Disposition: session Content-Length: 261 Remote-Party-ID: "Outbound Call" <sip:sean@ A.B.C.D >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1995227603 1995227604 IN IP4 A.B.C.D s=FreeSWITCH c=IN IP4 A.B.C.D t=0 0 m=audio 24636 RTP/AVP 0 19 101 c=IN IP4 A.B.C.D a=rtpmap:0 PCMU/8000 a=rtpmap:19 CN/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 This is not the case when the call direction is reversed. Like I said, the only difference appears to be that one direction is responding with a 183 while this one is responding with a 200. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org