[asterisk-dev] Insight into smoother and how/why asterisk repacks 60msec rtp stream to 20 when bridging

[Stephen Davies]

Hi,

I'm looking for any pointers to help me understand Asterisk rtp.c and the
handling of SIP-to-SIP bridging for audio.

The background:   I'm having trouble because one of my termination suppliers
has started sending RTP streams with 60msec of audio in each RTP packet.

Our Asterisk boxes are bridging that SIP stream out to another SIP and in
the process changing the packetisation from 60msec to 20msec.

The problem with that being that this results in 3 RTP packets being sent in
a "bunch".  The network between the server and the phone is resequencing
those packets, the SNOM320 phone at the edge is not coping with that at all
well and the call sounds bad.

I'm trying to get the network fixed, but this is an obscure point and these
days its hard to get your hands on someone at the ISP who can even
understand the problem, never mind fix it.

In the meantime the problem will be avoided if I could persuade Asterisk to
"follow-the-leaders" and maintain the 60msec packetisation on the way out.

I did find doc/rtp-packetization.txt and I'm going to experiment with
autoframing=yes and the other stuff, but I'd appreciate any other hints or
pointers to documentation to understand this better.

Thanks.
Steve
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