[Asterisk-Dev] RTCP-support

[Filip Olsson]

Hi there,

I've written a patch that adds some support for RTCP in Asterisk.

Some gateways require that the other end send RTCP-packets during
the session or the call will be dropped after a couple of minutes or
seconds.

This patch has been tested against a number of different SIP
useragents and gateways including Ciscos 79xx, AlliedTelesyn RG6xx,
AudioCodes MP-series, 42Networks, Granstream, X-Lite/X-Pro, Cisco
gateways and VocalTec gateways.

Some of those clients/servers require that we send these
RTCP-reports, the ones that didn't drop the calls before you applied
this patch shouldn't drop them after you apply it either =)

If you experience dropped calls at regular intervals this patch might be the solution.

The patch adds a CLI command to 'debug' RTCP-transmissions so you
can see the contents of the reports. For those of you that don't know
what they contain, apply the patch and you'll see =)

Please apply the patch and test it, you'll find more information at:

http://bugs.digium.com/view.php?id=2863

There is also a patch available for chan_sip that puts the
accumulated stats for the session into the userfield of the CDR upon
hangup, it's also in 2863.

If you report your results there's a big chance I'll fix them.

//Filip