[Asterisk-Dev] RTCP-support

[Olle E. Johansson]

Filip Olsson wrote:
> 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,
Thanks for this work! The rest of you: PLEASE test this now, so we can
decide whether it's ready for 1.2 or not.

/O