[Freeswitch-users] rtpmap line missing on answer

[David Ponzone]

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

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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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