Debugging RTP packets sending

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Am 24.03.2010 18:52, schrieb Olle Frimanson:
> Hi,
>
> I formulated my self wrongly, I meant sending multiple voice frames per RTP
> packet (ptime>  20 ms)

Ok. Probably I misunderstood you.

> The RTP/UDP/IP overhead is ~16 kbit/s for a single frame/pkt so for a low
> bit rate codec that is ~25 kbit/s and few GPRS links have that.

True.

> Secondly latency is totally random from low ~50 ms to really high could be
> seconds in worse case.
>
> This at least our experience.

50ms? I guess this is a rather new UTMS link.

The high delay is really a problem. I think this also relates to the 
sending buffer and retransmissions on layer 2. The GPRS network layer 
does not know that this packet is RTP (which means it is better to drop 
it instead of retransmitting and causing a traffic jam on the link)

regards
klaus



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