I've added FS-4382, please check there for further info.

/Peter

-----Ursprungligt meddelande-----

Från: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Peter Olsson

Skickat: den 3 juli 2012 18:07

Till: freeswitch-dev@lists.freeswitch.org

Ämne: Re: [Freeswitch-dev] Question about RTP marker

Thank you for your replies, both Tony and Kristian - it made things a little bit clearer for me :)

I will submit a proposed patch to Jira later on (probably tomorrow), and you can review my changes there, it's a small change.

/Peter

-----Ursprungligt meddelande-----

Från: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] För Kristian Kielhofner

Skickat: den 3 juli 2012 17:33

Till: freeswitch-dev@lists.freeswitch.org

Ämne: Re: [Freeswitch-dev] Question about RTP marker

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@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@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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