[Freeswitch-users] RTP Timestamp changes after REFER - Question on RFC conformity

[Peter Olsson]

Also, in current git head it might be possible that the RTP bug CHANGE_SSRC_ON_MARKER helps. It will change to a new SSRC every time the marker is sent. However, I didn't look into the pcaps, so it's possible that it's not enough for this specific case.

/Peter

-----Ursprungligt meddelande-----
Från: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] För Anthony Minessale
Skickat: den 23 augusti 2012 19:04
Till: FreeSWITCH Users Help
Ämne: Re: [Freeswitch-users] RTP Timestamp changes after REFER - Question on RFC conformity

The problem here is you can easily blur the lines on what is correct or incorrect and a possible fix might even break other things.

http://sidious.freeswitch.org/patches/rtp_reset.diff

Try this patch and see if its any better.  It will send a mark and new SSRC every time a channel is in a new bridge situation.





On Thu, Aug 23, 2012 at 10:31 AM, Christian Benke <benkokakao at gmail.com> wrote:
> Hi!
>
> Here's a trace: http://poab.org/misc/one_way_audio_internal_g711.pcap
> (or http://filedump.org/files/yMnIRQUC1345735448.html for the archive)
>
> The REFER happens at Paket 1305, Time 25.662729
>
> The RTP-Paket that are sent to the C-endpoint after the REFER are 
> continuous from A's point of view(As it is the continuation of the 
> A-B-stream), but not from C's point of view.
>
> Regards,
> Christian
>
> On 23 August 2012 17:18, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>> Do you have a pcap of it not setting the mark bit when the timestamp 
>> changes because the code will always set the mark bit when the packet 
>> is about to send is not exactly the next packet it expected to send 
>> after the previous one/
>>
>>
>> On Thu, Aug 23, 2012 at 10:04 AM, Christian Benke <benkokakao at gmail.com> wrote:
>>> Hi!
>>>
>>> I had a problem with VOP Softclient(www.voiceoperatorpanel.com)  
>>> where the callee had one-way-audio after a call was REFERred to him. 
>>> After lots of tracing, debugging and hairpulling i realized the 
>>> problem lies in the changing rtp-timestamp when the RTP-stream is 
>>> switched from the middleman to the initial call after the REFER. The 
>>> Softclient was not able to cope with the changed timestamp and 
>>> ignored the incoming RTP-packets, leading to no audio for the callee.
>>>
>>> I was eventually able to solve the problem by activating 
>>> rtp-rewrite-timestamps on the profile(Also added it to 
>>> http://wiki.freeswitch.org/wiki/RTP_Issues#Voiceoperatorpanel_VOP).
>>>
>>> However, i would like to know if FreeSWITCH/Sofia is working 
>>> according to the RFC and if the Softclient is to blame for the 
>>> problem(So i can file a bugreport with them).
>>>
>>> In this thread, Brian West states that  it's ok to skip forward in 
>>> timestamps as long as the marker-bit is set:
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060
>>> 333.html However, the Marker-bit is not set by FreeSWITCH when the 
>>> REFER occurs.
>>>
>>> I didn't find this stated in http://tools.ietf.org/html/rfc3550(But
>>> "timestamp" is mentioned a lot, so i may have missed it) but there's 
>>> this bug-report for Asterisk, where the exact same problem is 
>>> described and eventually handled:
>>> https://issues.asterisk.org/view.php?id=17007
>>>
>>> Could someone with more insight please elaborate?
>>>
>>> Best regards,
>>> Christian
>>>
>>> ____________________________________________________________________
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>>>  
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>>
>>
>>
>> --
>> Anthony Minessale II
>>
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>>
>>  
>> 
>>
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>
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>  
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--
Anthony Minessale II

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