[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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