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. Thanks, Sammy On Thu, Feb 16, 2012 at 10:00 PM, Kevin P. Fleming <[email protected]>wrote: > On 02/16/2012 01:16 AM, Sammy Govind wrote: > >> Hello list, >> >> I need to know about Asterisk's friendly nature with RTCP. I've phones >> which support RTCP and they connect to the outer world via multiple >> carriers. In one of my recent packet traces I've observed that the >> caller initiated a call with rtcp string in SDP while for the same >> call dialling our from Asterisk to the carrier has no RTCP string in SDP ! >> Can anyone please tell why is this so! or if there is anything I can do >> to make RTCPs flow through the asterisk server ! >> I've asterisk 1.6.2.20 in production. >> > > 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. > > RTCP will never *flow through* Asterisk, as Asterisk is terminating both > RTP flows and thus is an endpoint for both of them. > > Do you have an actual problem you are trying to resolve, or are you just > asking questions about RTCP? > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: [email protected] | SIP: [email protected] | 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<http://lists.digium.com/mailman/listinfo/asterisk-users> >
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