Re: [Linphone-developers] oRTP scheduler too fast??

[Simon MORLAT]

Hello,

With ortp-0.6.3preX you don't need glib at all.
Looking at the ethereal log (very interesting thing), you can see that 
rtp packets are sent by group of 3 or 4 nearly at the same time. They 
are not sent at regular interval.
This happens very often, mostly because of bad soundcard/sound drivers 
that can't grab small buffers of sound. So the rtp layer has to sent 
them by bursts.
But the receiver may be upset of that.
try calling rtp_session_set_jitter_compensation(rtpsession,100);
100 means 100 miliseconds, this is the time of bufferized audio before 
it is delivered to your application.
Your rtp sender seems to send 60-80ms of audio simultaneously, so a 
jitter compensation of 100ms should be enough for the receiver.
Having an automatic jitter-compensation setting is a direction for a 
future work...
Simon

Tomas Lundqvist wrote:

> Ok, so I don't need glib at all then? Not when copiling ortp and not when 
> compiling my application?
>
> I have added SIP functionality to rtprecv.c (ortp-0.6.3pre4). You can see the code in "ortpsiptest.c". I tested it both against Linphone and the windowsprogram Ubiquity Helsman UA (www.sipcenter.com) and I get the same bad result. You can see the output for this in recvtestout. I compiled both with and without glib. 
> You can compare the result with the output from my app in "myappstat". The statistics are just as bad for both apps. 
>
> I have investigated the SIP/RTP-traffic with "Ethereal" and it seems perfectly fine to 
> me. You can see the traffic (Helsman UA) in "recvtest".
>
> So now I have tested two different apps, two different ortp versions, two 
> different sending SIP UA running on two different OS and I have tried all 
> different combinations of these, and the oRTP API is always unsyncronized with 
> the incoming RTP stream. Additionally I have tried to compile the apps with and 
> without glib. The result is always the same.
>
> If you look at the Ethereal output you will see that the RTP packets contains 
> 160 bytes of G.711 ALAW. And that the timestamp continously increases with 160. 
> Aso, there are no missing packets and the packets are sent with continous time 
> intervals (nice 100Mbit LAN conditions).
>
> Would there bee any difference if I send from this computer and receive at the 
> other? (same Linux distro but intel PIII instead of AMD Athlon)
>
> I just don't know what to do! Shouldn't it be possible to receive an RTP stream 
> from a SIP UA (Linphone). I can't see that there should be anything wrong with 
> linphone since the result is the same with Helsman UA.
>
> I really like your API and I want to use it!
>
> Regards,
>
>  
>
>
> --
> *****************************
> TOMAS LUNDQVIST
> www.tomaslundqvist.com
> address@hidden
> *****************************
>
> --
>
>
>