Debugging RTP packets sending

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Hi,

I formulated my self wrongly, I meant sending multiple voice frames per RTP
packet (ptime > 20 ms)

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.

Secondly latency is totally random from low ~50 ms to really high could be
seconds in worse case.

This at least our experience.

BR/Olle


-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@xxxxxxxxx] 
Sent: den 24 mars 2010 18:13
To: pjsip list
Cc: Olle Frimanson
Subject: Re: Debugging RTP packets sending

Hi Olle!

Am 24.03.2010 17:53, schrieb Olle Frimanson:
> The I'm just querious how you should get PJSIP work on GPRS, all the
> test we have done shows this is very hard in rel conditions even if you
> send multiple packets per frame, especially on the uplink.

Sending multiple packets can be counter productive. The GPRS devices I 
tested with did have a huge sending buffer, and GPRS has retransmissions 
on layer2. So, packet loss on the mobile link is really rare - also with 
bad connectivity.

By sending packets twice you need double bandwidth and if the needed 
bandwidth is not available, due to the device's sending buffer the delay 
increases.

I once did lots of T.38 faxing over GPRS. Everybody told us that we have 
an unreliable link thus we have to enable T.38 redundancy - but that was 
the totally wrong approach (as described above). Yes, GPRS is slow, but 
GPRS is rather reliable. Thus, once we turned-off T.38 redundancy we 
were able to fax over GPRS.

regards
Klaus

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