Rtp/g711 packages

[Moriquendi]

Hello,

I am trying to send rtp/g711 packages with vlc to an ip telephony. The audio I am experiencing has intermittent sound, like a robotic sound. In order words, its like you hear 1st,3rd,5th,.. packages and 2nd,4th,6th,.. packages are lost/cant be heard. When I try the same setup on another pc, its just fine, however, the ip telephony testing seems to be problematic. Wireshark results shows that there is either 371ms or (mostly)743 ms between two g711 packages. Honestly, I am not sure what to do to fix this issue, however, I am thinking about inreasing the frame sizes of packages or the streaming frequency of rtp/g711 packages.

Can you tell me if I am heading to a correct direction, if so how can I implement it inside a transcoded stream such as;

Code: Select all

transcode{vcodec=none,samplerate=8000,acodec=alaw,ab=16}:duplicate{dst=rtp{dst=230.0.0.0,port=8004,sdp=sap://,name=TestStream}}"

Thanks in advance.

[Rémi Denis-Courmont]

Does the stream sound correct if you open it in VLC (rtp://@230.0.0.0:8004) ?

[Moriquendi]

Does the stream sound correct if you open it in VLC (rtp://@230.0.0.0:8004) ?

yup, it sounds quite alright. The wireshark logs when i listen it on (rtp://@230.0.0.0:8004) look the same (743 ms delays between packages, same Rtp format etc.)

[Rémi Denis-Courmont]

So then it's probably a limitation on your receiver. It might not like the packet size or the delay jitter. Unless you can tune the receive, I don't think there is anyway to fix this other than digging into the VLC source code.

[Moriquendi]

On a side note, for the successfull communications between two IP telephones I am using, there are 160 ms delays between two consecutive audio(g711) packages. I am not sure if this has anything to do with our issue.

[Rémi Denis-Courmont]

The most standard value are 20ms and 50ms, I think.