[Freeswitch-dev] Question about RTP marker
anthony.minessale at gmail.com
Tue Jul 3 19:18:24 MSD 2012
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
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
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Anthony Minessale II
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