<div dir="ltr"><div>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.<br></div><div><div>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.<br></div><div>I'm I wrong? (it's sooome time since I've done C code ;-)<br></div><div><br></div><div>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:</div><div><br><div><font face="monospace, monospace">2016-05-23 08:59:41.252648 [INFO] switch_core_media.c:6549 Activating RTCP PASSTHRU PORT 0</font></div><div><font face="monospace, monospace">2016-05-23 08:59:41.252648 [DEBUG] switch_rtp.c:4189 RTCP passthru enabled. Remote Port: 29991</font></div></div><div><br></div><div>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)</div><div><br></div><div>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? <br></div><div><br></div><div>Hope someone can help ;-)<br></div><div> </div><div>Regards,</div><div> Allan</div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Sat, May 21, 2016 at 7:13 PM, Allan Kristensen <span dir="ltr"><<a href="mailto:ak@hejdu.dk" target="_blank">ak@hejdu.dk</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hello,<div><br></div><div>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 ;-)</div><div><br></div><div>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).</div><div><br></div><div>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.</div><div><br></div><div>I've added the "rtcp-audio-interval-msec" to the sip profile, so it should work, but it's not..</div><div>I've checked the source and even enabled the RTCP_DEBUG in the source to get a clue, but no help there.</div><div><br></div><div>Asterisk is not actually adding anything about rtcp to the sdp, but is that really needed to get the rtcp working?<br></div><div>I'm not using ICE or anything, just a simple setup for this.</div><div><br></div><div>Is there any way to see if FS is disabling the rtcp feature and if so, why?</div><div><br></div><div>Any ideas on what can be done to get things working? ;-)</div><div><br></div><div>Thank you and have a excellent day!</div><span class="HOEnZb"><font color="#888888"><div>/ Allan</div><div><br></div><div><br></div></font></span></div>
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