Re: [asterisk-users] Asterisk RTCP
Hi kevin, 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 A-leg===Asterisk and Asterisk===B-leg 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 !! Please suggest. Regards. Sammy On Fri, Feb 17, 2012 at 5:21 PM, Gohar Ahmed gohar.ah...@vopium.com wrote: 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 -- _ -- 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
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
Re: [asterisk-users] Asterisk RTCP
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
Re: [asterisk-users] Asterisk RTCP
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 kpflem...@digium.comwrote: 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: 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-**usershttp://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
