[Freeswitch-dev] Question about RTP marker

[Kristian Kielhofner]

+1

The use of the RTP marker has been broken and misunderstood by many
implementations...

The references the OP found about "talkspurts" are probably referring
to the use of markers with VAD (voice activity detection).  When VAD
detects speech the first RTP packet is supposed to have the marker
set, almost as a way of saying "Hey! I'm talking again".  Of course
with VAD a new talkspurt will also have a different timestamp...

FreeSWITCH (out of respect for jitter buffers, as Anthony said) tries
to pass original (read: meaningful) timestamps and use markers
wherever possible.  Consider the following scenario:

Call -> IVR (FS starts generating RTP)
IVR -> bridge to external SIP device (device starts generating RTP)

In this scenario FS will generate the initial RTP stream with it's own
timestamps.  When the call is bridged to the remote device FS will set
the marker and pass the timestamps from the B leg to the A leg.  This
allows the far end jitterbuffers (not FS) to buffer jitter based on
the actual end to end RTP stream.

Unfortunately this can cause all kinds of confusion with broken implementations.

While changing the SSRC is (of course) a hack if it improves
compatibility while maintaining the usefulness and integrity of
timestamps (and jitter buffers) I'm for it.

On Tue, Jul 3, 2012 at 11:18 AM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> The line is really blurry between what is right and what works better.
> I think the only real rule about the marker bit is that you are
> supposed to set it any time the timestamp is not the next expected
> timestamp.
>
> This is supposed to help you to understand that you are purposely
> sending a packet that has a different timestamp base or that you are
> resuming after intentionally not sending packets for a period of time.
>  This tells the remote end to flush its jitter buffer which probably
> explains the drop in audio.
>
> If changing the SSRC is an acceptable workaround for when the rtp
> streams change that do not result in audio gaps, I'm more than happy
> to accept it as a default and we can always make it a config option if
> it presents any problems.
>
>
>
> On Tue, Jul 3, 2012 at 9:39 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
>> When I’ve been debugging RTP streams in Wireshark, I’ve noticed that
>> Whireshark will get “everything wrong” when the timestamp changes (is
>> reset). For instance, if I call a conference, by the time I get into the
>> conference, the RTP timestamps will start over, and FS will send the marker
>> bit By that time Wireshark will report strange figures for jitter etc.
>>
>>
>>
>> After some Googling I’ve found that the meaning of the RTP marker is to
>> indicate “the beginning of a talkspurt”. Which is not very clear to me... :)
>>
>>
>>
>> So my question is – is RTP marker really enough when starting a “new RTP
>> stream” (not really new, but since the timestamps are reset we can think of
>> it like that) – according to what I’ve been able to find, maybe the SSRC
>> should also be changed?
>>
>>
>>
>> I have created a patch that will change the SSRC when a reset like this
>> occurs, and after that Wireshark detects my audio streams without problems
>> (though, it will detect more streams), but no strange jitter figures etc.
>> are reported.
>>
>>
>>
>> The reason I started digging into this was because a PBX I was connecting to
>> lost about the first second of audio after connecting to the conference,
>> after also changing the SSRC the problem seems resolved.
>>
>>
>>
>> I could make a RTP_BUG setting for this, but I’m not really sure if I should
>> really care – that’s why I’m asking here, if anyone has more knowledge about
>> this?
>>
>>
>>
>> /Peter
>>
>>
>>
>>
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>
>
>
> --
> Anthony Minessale II
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>
> 
> 
>
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>
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-- 
Kristian Kielhofner