Hello list, Kevin I agree with you on independent monitored entity for A leg while the outbound leg has separate QoS measures. But after this thread I went to my monitoring tool and saw that for some calls on the same asterisk setup I had no RTP or RTCP while there were calls with both RTP and RTCP captured as well.
Since I've a SIP proxy on top of asterisk servers layers, could it be possible that RTP and RTCPs bypass asterisk (media redirect) and that's why I see RTCPs and RTPs logged into monitoring tool while those call who couldn't redirect/bypass media from asterisk don't show any RTCPs!? Sammy can you provide further details of your setup please! Regards, Gohar -----Original Message----- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Friday, February 17, 2012 5:02 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk && RTCP On 02/17/2012 12:09 AM, Sammy Govind wrote: > Hello, > > 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. > > 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. As I said before, you will never get RTCP *flowing through* Asterisk. When your softphone calls Asterisk, that will be a separate call leg from the one from Asterisk to your provider. Your monitoring tool should treat those as separate call legs and produce an analysis for them independently. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users