[Freeswitch-users] Can't get rtcp to work...

[Allan Kristensen]

Ok, digged into 'switch_core_media.c' and apparently the 'remote_rtcp_port'
is 0, so the function 'switch_rtp_activate_rtcp' is never called, which I
assume means rtcp is never enabled.
>From what I can see, it's set from channel variable
"rtp_remote_audio_rtcp_port" and going backwards it looks like it comes
from negotiating the sdp and the "a=rtcp:<port>" - so because the remote
sip client never provides this, nothing is set.
I'm I wrong? (it's sooome time since I've done C code ;-)

Just to make sure I did a <param name="rtcp-video-interval-msec"
value="passthru"/> in the sip_profile because I can see 'remote_port'
branching happen later in the source and voilá:

2016-05-23 08:59:41.252648 [INFO] switch_core_media.c:6549 Activating RTCP
PASSTHRU PORT 0
2016-05-23 08:59:41.252648 [DEBUG] switch_rtp.c:4189 RTCP passthru enabled.
Remote Port: 29991

However I wish to monitor each leg separately, so this is not a viable
option to me (and I'm no sure it works with remote port = 0 actually,
haven't checked the code)

Long story short, couldn't/shouldn't FS just assume that 'remote_rtcp_port'
is rtp_port +1 if 'a=rtcp' is missing in the in the SDP?

Hope someone can help ;-)

Regards,
  Allan

On Sat, May 21, 2016 at 7:13 PM, Allan Kristensen <ak at hejdu.dk> wrote:

> Hello,
>
> I've recently moved from Asterisk to Freeswitch and I'm trying to setup a
> FS on a cloud solution. I have mixed results from previous projects with
> Asterisk on cloud servers, but this one is just going to proxy media, so
> I'll give it a shot ;-)
>
> I'm going to log the rtcp stats into the cdr, so I can be sure things are
> ok, but most of the rtcp related stats all contain "0" and mos score 4.5
> (even if I randomly drop rtp packets for a couple of seconds to simulate
> packet loss... not good).
>
> The client uses an asterisk, which is sending rtcp reciever reports to FS,
> however FS is not listening on the rtcp port at all (rtp port +1) and
> server just sends back icmp unreach. (it's not muxed into the rtp) Also FS
> is not sending rtcp packets, from what I can see, so I suspect rtcp is
> somehow not enabled for the channel / session at all.
>
> I've added the "rtcp-audio-interval-msec" to the sip profile, so it should
> work, but it's not..
> I've checked the source and even enabled the RTCP_DEBUG in the source to
> get a clue, but no help there.
>
> Asterisk is not actually adding anything about rtcp to the sdp, but is
> that really needed to get the rtcp working?
> I'm not using ICE or anything, just a simple setup for this.
>
> Is there any way to see if FS is disabling the rtcp feature and if so, why?
>
> Any ideas on what can be done to get things working? ;-)
>
> Thank you and have a excellent day!
> / Allan
>
>
>
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