[Freeswitch-users] rtpmap line missing on answer

[Steven Ayre]

Who is answering the call? Are you sure they're not choosing a different
codec from the ones offered in the INVITE?

-Steve


On 23 May 2011 16:52, Sean Eichhorn <seichhorn at gci.com> wrote:

>  Yes, I understand that this is the Cisco proprietary codec.  The problem
> is that between two Cisco systems the SIP endpoints will not use this
> protocol if it is not sent in the SDP.  The **original** SDP does contain
> the codec, but when the call is answered, another SDP is sent.  This SDP is
> altered by Freeswitch and all codecs other than the selected codec, NTE, and
> comfort noise are removed.
>
> It might not be Freeswitch doing it.  I’m digging into the source and it
> looks like it might be the Sofia-SIP library.
>
> The funny thing is that it works in one direction, but not the other.
>
> FS subscriber -> CIDR defined static SIP endpoint (WORKS!)
>
> CIDR defined SIP endpoint -> FS Subscriber (Doesn’t work)
>
>
>
> It seems that one of the major differences is that the subscriber answers
> with a 200 OK, whereas the CIDR endpoint answers with a 183.
>
>
>
> I’m currently searching to see if there is a way to change the response
> type, but so far no luck.
>
>
>
> Sean Eichhorn
>
> General Communications Inc.
>
> IP Telephony Engineer
>
> (907) 868-6902
>
>
>
> -No telephony products were harmed in the making of the message-
>
> *From:* David Ponzone [mailto:david.ponzone at ipeva.fr]
> *Sent:* Friday, May 20, 2011 01:03
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] rtpmap line missing on answer
>
>
>
> Sean,e
>
>
>
> X-NSE is the Clear Channel codec from Cisco.
>
> I suspect FreeSWITCH does not support it, and as such, it can't be offered
> to leg B.
>
> Perhaps there is a way to enable in outbound codecs as bypass, but I really
> doubt so.
>
> Though, you could try to enable late-negotiation where client and gateway
> will negotiate together.
>
>
>
> 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 20/05/2011 à 20:54, Sean Eichhorn a écrit :
>
>
>
>   I have a freeswitch system that (for the most part) works flawlessly for
> me.  However, I need to have an additional rtpmap line in the SDP when a
> call is answered.
>
> My client sends the following SDP upon answering the call :
>
>    m=audio 17214 RTP/AVP 0 19 101 100
>
>    c=IN IP4 192.168.98.79
>
>    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
>
>
>
> Freeswitch forwards the following SDP to the external gateway :
>
>    m=audio 20654 RTP/AVP 0 19 101
>
>    c=IN IP4 66.223.187.208
>
>    a=rtpmap:0 PCMU/8000
>
>    a=rtpmap:19 CN/8000
>
>    a=rtpmap:101 telephone-event/8000
>
>    a=fmtp:101 0-16
>
>
>
> The other information appears fine.  Everything works, but my client-side
> is not receiving the a=rtpmap:100 line.
>
> This issue appears regardless of whether or not I’m in proxy mode, bypass
> mode, or neither.
>
>
>
> Using "sip_append_audio_sdp” has no effect on the answering SDP, only the
> initial offer.
>
>
>
> Any ideas?
>
>
>
> Thanks in advance,
>
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