[Freeswitch-users] rtpmap line missing on answer
David Ponzone
david.ponzone at ipeva.fr
Wed May 25 01:12:01 MSD 2011
Sean,
can you check the configuration of the SIP profile used by both legs ?
Perhaps you have late-negotiation enabled in one of them ?
David Ponzone Direction Technique
email: david.ponzone at 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 - 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 18:31, Sean Eichhorn a écrit :
> 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 at A.B.C.D>;tag=yy2XQKXSy7gtN
> To: <sip:xxxxx02 at 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 at I.J.K.L>;party=called;screen=no;privacy=off
> Contact: <sip:xxxxx02 at 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 at A.B.C.D>;tag=65BCD9B0-115E
> To: <sip:xxxxx02 at A.B.C.D>;tag=XN94Nrcp1yt7S
> Call-ID: A4A3C9E0-855A11E0-8237C79A-B2A2FA42 at E.F.G.H
> CSeq: 102 INVITE
> Contact: <sip:xxxxx02 at 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 at 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 at lists.freeswitch.org [mailto:freeswitch-users-bounces at 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 at 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 - 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 at A.B.C.D>;tag=NH1HrNe744HSB
> To: <sip:sean at 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 at 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 at 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.
>
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