Hi kevin,<div><br><div>I've observed that I've "rtcp set debug" command (rtcp based commands) available on my asterisk console. Can you please explain about RTCP. I really need RTCPs in my setup, it doesnt matter if the RTCPs are separate for both A-leg and B-leg i.e </div>
<div>A-leg<===>Asterisk</div><div>and </div><div>Asterisk<===>B-leg</div><div>I can live with RTPs flowing for each leg with asterisk separately. But problem is I dont get any RTCPs for each leg independently as well !!</div>
<div><br></div><div>Please suggest.</div><div><br></div><div>Regards.</div><div>Sammy<br><div><div><br><div class="gmail_quote">On Fri, Feb 17, 2012 at 5:21 PM, Gohar Ahmed <span dir="ltr"><<a href="mailto:gohar.ahmed@vopium.com">gohar.ahmed@vopium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello list,<br>
<br>
Kevin I agree with you on independent monitored entity for A leg while the<br>
outbound leg has separate QoS measures. But after this thread I went to my<br>
monitoring tool and saw that for some calls on the same asterisk setup I had<br>
no RTP or RTCP while there were calls with both RTP and RTCP captured as<br>
well.<br>
<br>
Since I've a SIP proxy on top of asterisk servers layers, could it be<br>
possible that RTP and RTCPs bypass asterisk (media redirect) and that's why<br>
I see RTCPs and RTPs logged into monitoring tool while those call who<br>
couldn't redirect/bypass media from asterisk don't show any RTCPs!?<br>
<br>
Sammy can you provide further details of your setup please!<br>
<br>
Regards,<br>
Gohar<br>
<div class="HOEnZb"><div class="h5"><br>
-----Original Message-----<br>
From: <a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a><br>
[mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of Kevin P.<br>
Fleming<br>
Sent: Friday, February 17, 2012 5:02 PM<br>
To: <a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a><br>
Subject: Re: [asterisk-users] Asterisk && RTCP<br>
<br>
On 02/17/2012 12:09 AM, Sammy Govind wrote:<br>
> Hello,<br>
><br>
> Thanks for taking out tome for my query. Yes I do have an actual<br>
> problem. I've a monitoring tool to record the VoIP QoS (Asterisk servers<br>
> port mirrored to it). My end points(soft-phones) are sending RTCP<br>
> connection strings to asterisk, and Asterisk then forwards their call to<br>
> their destination choosing any suitable carrier.<br>
><br>
> If I don't get RTCP flowing through asterisk the monitoring tool simply<br>
> fails to display and call stats. Please advice what should I be doing to<br>
> cater this.<br>
<br>
As I said before, you will never get RTCP *flowing through* Asterisk.<br>
When your softphone calls Asterisk, that will be a separate call leg<br>
from the one from Asterisk to your provider. Your monitoring tool should<br>
treat those as separate call legs and produce an analysis for them<br>
independently.<br>
<br>
--<br>
Kevin P. Fleming<br>
Digium, Inc. | Director of Software Technologies<br>
Jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a> | Skype: kpfleming<br>
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