Hello,<div><br><div>Thanks for taking out tome for my query. Yes I do have an actual problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers port mirrored to it). My end points(soft-phones) are sending RTCP connection strings to asterisk, and Asterisk then forwards their call to their destination choosing any suitable carrier.</div>
<div><br></div><div>If I don't get RTCP flowing through asterisk the monitoring tool simply fails to display and call stats. Please advice what should I be doing to cater this.</div><div><br></div><div>Thanks,</div><div>
Sammy</div><div><br><div class="gmail_quote">On Thu, Feb 16, 2012 at 10:00 PM, Kevin P. Fleming <span dir="ltr"><<a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div class="HOEnZb"><div class="h5">On 02/16/2012 01:16 AM, Sammy Govind wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hello list,<br>
<br>
I need to know about Asterisk's friendly nature with RTCP. I've phones<br>
which support RTCP and they connect to the outer world via multiple<br>
carriers. In one of my recent packet traces I've observed that the<br>
caller initiated a call with rtcp string in SDP while for the same<br>
call dialling our from Asterisk to the carrier has no RTCP string in SDP !<br>
Can anyone please tell why is this so! or if there is anything I can do<br>
to make RTCPs flow through the asterisk server !<br>
I've asterisk 1.6.2.20 in production.<br>
</blockquote>
<br></div></div>
It is not mandatory to signal anything related to RTCP in the SDP. RTCP is implicitly handled on the next port up from the port being used for RTP; the signaling in SDP is only needed if the RTCP is *not* going to be on the next port up.<br>
<br>
RTCP will never *flow through* Asterisk, as Asterisk is terminating both RTP flows and thus is an endpoint for both of them.<br>
<br>
Do you have an actual problem you are trying to resolve, or are you just asking questions about RTCP?<br>
<br>
-- <br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href="mailto:kfleming@digium.com" target="_blank">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com" target="_blank">kpfleming@digium.com</a> | Skype: kpfleming<br>
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