[asterisk-dev] RTP timestamp problems or other?

[Nick Lewis]

>By sending audio form a file in 20ms 
>blocks over RTP on the receiver the stram 
>will be played too fast. If I´m doing a 
>pause of 20ms before sending
>each packet, it will be run good. 
>Is this problem on the timestamp?

It is just a guess as I am not familiar with the rtp handling specifically in asterisk but you may be expecting too much of any buffering functionality it may have. Even if asterisk wanted to play out the audio as per the timestamps, its jitter buffer would repeatedly overflow and be resynched if the incoming rtp frames were being received at much above realtime. This would not be a weakness in asterisk as there is good reason to keeping buffering to a minimum. Latency is much more important than delivery in these sorts of comms.

Irrespective of the other equipment, having the softphone pause before sending each packet seems the correct implementation.

-- N_L

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