[Asterisk-Dev] RTCP-support

[Dan Evans]

A question regarding this:

We have a user agent that sends RTCP packets, but we had to turn that 
off when talking to Asterisk.  We found that Asterisk would shift its 
RTP port in midstream to that used by RTCP (RTP+1), after it received an 
RTCP packet. Did you find this to be a problem?

Dan Evans

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
> 
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