[asterisk-users] Asterisk && RTCP

Sammy Govind govoiper at gmail.com
Fri Feb 17 00:09:22 CST 2012


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 <kpfleming at digium.com>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: kfleming at digium.com | SIP: kpfleming at digium.com | Skype: kpfleming
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> Check us out at www.digium.com & www.asterisk.org
>
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