[pjsip] RTP Silence Suppression

Nanang Izzuddin nanang at pjsip.org
Tue Apr 22 10:00:55 EDT 2008


Hi Thomas,

"Removing A from the B's conference bridge" is a bit ambiguous
sentence though :)
Assumed in the B side, the conference bridge only has 2 ports (sound
port is port 0 and stream port is port 1) then issuing command:
- "cd 0 1" will make B stop sending RTP to A (this will save traffic,
but NAT session may be gone)
- "cd 1 0" will not save traffic.
These scenarios don't seem to involve silence suppression.

You can test the silence suppression by not talking or muting mic (to
make it safe from keyboard stroke sound :D), then you can check the
traffic by using command 'dq' periodically in pjsua. Please make sure
the VAD is enabled (just don't put --no-vad in the pjsua's param).


Regards,
nanang


On 22/04/2008, Thomas Plotkowiak <plotti at gmx.net> wrote:
> Scenario with Psjua:
>
> A initiates SIP call with B.
>
> B adds A to the conference bridge.
> B is able to hear A.
> --> In RTP protocoll audio is transmitted from A to B.
>
> What if i now remove A from the conference bridge and still have the call.
>  Does this lead to silence supresssion on B or A side? Do I save traffic by
> doing this?
>
> How could I prove that I am saving traffic with this? Can I see it somewhere
> happening in the logs?
>
> Any comments or info about this topic would be appreciated.
>
>
> Cheers
> Thomas
>
>
> RFC3389 on Silence Suppression: RTP allows discontinuous transmission
> (silence suppression) on any audio payload format. The receiver can detect
> silence suppression on the first packet received after the silence by
> observing that the RTP timestamp is not contiguous with the end of the
> interval covered by the previous packet even though the RTP sequence number
> has incremented only by one. The RTP marker bit is also normally set on such
> a packet.
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