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

[Anthony Minessale]

and when you do be sure to attach a full console log with sofia global
siptrace on enabled.


On Tue, Apr 26, 2011 at 10:02 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 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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>>
>
>
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>



-- 
Anthony Minessale II

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