The VideoLAN Forums

Hello again,

Is there an easy way to change the (rtp)packet sizes inside a transcoded stream or do I need to dig into the packetizer module in vlc?

Thanks.

Hmm, depends on the format. You can try to mess with the MTU setting but otherwise you need to edit the RTP packetization code.

Hi again,

As you suggested, I've tried couple of things with MTU but didnt seem to work for g711 packets. Afterwards, I made some experiments using audio files sampled at different frequencies (8k,11025,22050,44100 etc.).

Considering I have only g711(a-law) audio-codec available on the receiver side(SIP telephony), I am trying to encode the audio stream to g711 format only. From my wireshark results, I observed that timestamp delay between packets depend on the input stream's (not transcoded stream) sampling rate. Actually, for 8000 Hz, and 11025 Hz, they are same, and it gets higher as we increase the sampling rate. I assume this is due to the timestamp calculation of vlc packetizer module. Since my receiver side cant handle respectively large packets, I need to somehow make sure that timestamping or packetizing always stays under a desired value. I know you already suggested me to look into the source code of rtp packetizer, and I did, but can you be more specific which class I should look into (maybe direc.c?).

I have like 3 days to finish this project, thats why I am asking for further help.

Thanks.

p.s: For 8000 Hz and 11025 files, the sound is very good, rest sound very intermittent and robotic

I have never looked at this particular problem. I can't help you much further.