[Freeswitch-dev] One way audio problem from B-leg of the call on Session Refresh and HOLD if its a dynamic payload type

[Steven Ayre]

Open a jira for it so that the bug can be registered and tracked.

http://jira.freeswitch.org/

-Steve



On 26 April 2011 15:16, Jyotshna Cherukuri <jcherukuri_necc at yahoo.com>wrote:

> Hi ,
>
> I am working with the latest revision of Freeswitch and I am having one way
> audio problem  using SILK/8000 codec  on B-leg of the call after Session
> Refresh or on HOLD. This is due to the fact that when FS sends an INVITE out
> to B-leg it sends 98 in its SDP offer and B-leg responds back with same code
> but with payload type "120"
>
> [ Offer ]
> v=0
>    o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10
>    s=FreeSWITCH
>    c=IN IP4 192.168.2.10
>    t=0 0
>    m=audio 33766 RTP/AVP 98 9 101
>    a=rtpmap:98 SILK/8000
>    a=fmtp:98 useinbandfec=1;usedtx=0
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=silenceSupp:off - - - -
>    a=ptime:20
>
>
> [ Answer ]
>
>  v=0
>    o=- 3512815772 3512815773 IN IP4 192.168.4.121
>    s=pjmedia
>    c=IN IP4 192.168.4.121
>    t=0 0
>    a=X-nat:0
>    m=audio 4002 RTP/AVP 120 96
>    a=rtcp:4003 IN IP4 192.168.4.121
>    a=rtpmap:120 silk/8000
>    a=fmtp:120 useinbandfec=1;usedtx=0
>    a=sendrecv
>    a=rtpmap:96 telephone-event/8000
>    a=fmtp:96 0-15
>
> Freeswitch handles this properly on the initial offer/answer as its using
> this patch (tell rtp stack about what remote payload type to expect when
> the receiving end follows the stupid SHOULD as WONT and sends a different
> dynamic payload number than the one in the offer)
>
> After one min into the call because Session Timers are enabled Freeswitch
> sends a Session Refresh with payload type now setting to "120"
>
> [Refresh Offer]
> v=0
>    o=FreeSWITCH 1303773966 1303773967 IN IP4 192.168.2.10
>    s=FreeSWITCH
>    c=IN IP4 192.168.2.10
>    t=0 0
>    m=audio 33766 RTP/AVP 120 96 9
>    a=rtpmap:120 SILK/8000
>    a=fmtp:120 useinbandfec=1;usedtx=0
>    a=rtpmap:96 telephone-event/8000
>    a=fmtp:96 0-16
>    a=silenceSupp:off - - - -
>    a=ptime:20
>
> The remote end then starts sending RTP packets with payload number "120" in
> its RTP header and FS stops processing these packets and as a result is
> resulting in one-way audio issue.
>
> Any help is appreciated.
>
> Thanks in advance
> Regards
> Jyotshna
>
>
> P.S :  The issue starts even when the remote end presses" HOLD" as it sends
> INVITE on hold with "120" and FS responds back with "120" in its answer.
>
>
>
>
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