Does Asterisk support RTCP-XR in any way? I'm looking to use the quality monitoring now available on Polycom phones, but I can't figure out how to actually read the data. Everything I Google for is people asking and not receiving answers or articles about the "new" technology... 5 years ago. ---------- Mike Hammett Intelligent Computing Solutions http://www.ics-il.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080915/71a280dc/attachment.htm
At 5:37 PM -0500 2008/9/15, Mike Hammett wrote:> > >Does Asterisk support RTCP-XR in any way? I'm looking to use the >quality monitoring now available on Polycom phones, but I can't >figure out how to actually read the data. Everything I Google for >is people asking and not receiving answers or articles about the >"new" technology... 5 years ago. > > >---------- >Mike Hammett >Intelligent Computing Solutions >http://www.ics-il.comUnless a lot of code has snuck in without my seeing it, the answer right now is "No, it is not supported." However, that doesn't mean it wouldn't be a welcome addition. Your patches are awaited. :-) The RTCP-XR stuff is very interesting in that it gives (semi)real-time updates to all of the SIP elements in the signalling chain about the call quality that is happening in the RTP stream. This allows (potentially) for mid-call signalling components to redirect the calls to less-congested media servers, without having to sit in the RTP data path. Getting Asterisk or any call platform to do this would be complex. I think a better starting point would just be getting the call quality data back for an end-of-call quality record - it's a bit ambitious to do mid-call redirection at this point, I think, when there are more fundamental stats we're not getting or using. The RTCP-XR stuff has changed some of the data name and types in the last year or so, IIRC, so perhaps it's a good thing that it's been slow to show up in Asterisk. At my last conversation with someone who would know, the thinking was that it's getting very close to being approved and is mostly stable in the general sense of the protocol details. For a longer screed on Call Quality Detail Records, see links below. http://lists.digium.com/pipermail/asterisk-dev/2006-November/024586.html http://lists.digium.com/pipermail/asterisk-dev/2004-May/004180.html I have created minimalist CQDRs with whatever RTP data that is currently available within Asterisk, but I've yet to validate that they're getting "correct" data, and doesn't seem for the clients I've tested to report the remote side. Also, it seems that "remote_count" is a bit counter-intuitive - it says how many packets were sent to the remote side, not how many were received by the remote side. This is perhaps where the RTCP-XR stuff would be useful, or maybe we're just not fully exercising the RTCP data that we could. exten => 999,1,Playback(tt-monkeys) exten => 999,n,Log(DEBUG,local_ssrc: ${CHANNEL(rtpqos,audio,local_ssrc)}) exten => 999,n,Log(DEBUG,local_lostpackets: ${CHANNEL(rtpqos,audio,local_lostpackets)}) exten => 999,n,Log(DEBUG,local_jitter: ${CHANNEL(rtpqos,audio,local_jitter)}) exten => 999,n,Log(DEBUG,local_maxjitter: ${CHANNEL(rtpqos,audio,local_maxjitter)}) exten => 999,n,Log(DEBUG,local_minjitter: ${CHANNEL(rtpqos,audio,local_minjitter)}) exten => 999,n,Log(DEBUG,local_normdevjitter: ${CHANNEL(rtpqos,audio,local_normdevjitter)}) exten => 999,n,Log(DEBUG,local_stdevjitter: ${CHANNEL(rtpqos,audio,local_stdevjitter)}) exten => 999,n,Log(DEBUG,local_count: ${CHANNEL(rtpqos,audio,local_count)}) exten => 999,n,Log(DEBUG,remote_ssrc: ${CHANNEL(rtpqos,audio,remote_ssrc)}) exten => 999,n,Log(DEBUG,remote_lostpackets: ${CHANNEL(rtpqos,audio,remote_lostpackets)}) exten => 999,n,Log(DEBUG,remote_jitter: ${CHANNEL(rtpqos,audio,remote_jitter)}) exten => 999,n,Log(DEBUG,remote_maxjitter: ${CHANNEL(rtpqos,audio,remote_maxjitter)}) exten => 999,n,Log(DEBUG,remote_minjitter: ${CHANNEL(rtpqos,audio,remote_minjitter)}) exten => 999,n,Log(DEBUG,remote_normdevjitter: ${CHANNEL(rtpqos,audio,remote_normdevjitter)}) exten => 999,n,Log(DEBUG,remote_stdevjitte: ${CHANNEL(rtpqos,audio,remote_stdevjitter)}) exten => 999,n,Log(DEBUG,remote_count: ${CHANNEL(rtpqos,audio,remote_count)}) exten => 999,n,Log(DEBUG,maxrtt: ${CHANNEL(rtpqos,audio,maxrtt)}) exten => 999,n,Log(DEBUG,minrtt: ${CHANNEL(rtpqos,audio,minrtt)}) exten => 999,n,Log(DEBUG,normdevrtt: ${CHANNEL(rtpqos,audio,normdevrtt)}) exten => 999,n,Log(DEBUG,stdevrtt: ${CHANNEL(rtpqos,audio,stdevrtt)}) exten => 999,n,Log(DEBUG,rtpdest: ${CHANNEL(rtpdest,audio)}) exten => 999,n,Hangup JT -- John Todd jtodd at digium.com +1-256-428-6083 Asterisk Open Source Community Director
Hi, I fully agree that RTCP-XR support would be (more than) welcome in Asterisk but I was not aware of any SIP hardphone supporting it. Now, Googling with Polycom and RTCP-XR, I saw that RTCP-XR is included in Polycom Productivity Suite for Soundpoint. Is this Polycom Productivity Suite for Soundpoint an option (when buying a Polycom phone) ? Which other brand supports RTCP-XR ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080916/861f9ef3/attachment.htm
Another question : exten => 999,n,Log(DEBUG,local_ssrc:> ${CHANNEL(rtpqos,audio,local_ssrc)})Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an Asterisk version or is it something describing what should be coded ? Regards>-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080918/23d18ba4/attachment-0001.htm
Philipp Kempgen
2008-Sep-18 12:35 UTC
[asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)
Olivier schrieb:> Another question : > > exten => 999,n,Log(DEBUG,local_ssrc: >> ${CHANNEL(rtpqos,audio,local_ssrc)}) > > > Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in an > Asterisk versionYes. 1.4 and 1.6. But only for SIP channels obviously. chan_sip.c: acf_channel_read() Philipp Kempgen -- http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 --
Olivier
2008-Sep-18 14:41 UTC
[asterisk-users] CHANNEL((rtpqos|audio|local_ssrc)) (was: Re: RTCP-XR)
2008/9/18 Philipp Kempgen <philipp.kempgen at amooma.de>> Olivier schrieb: > > Another question : > > > > exten => 999,n,Log(DEBUG,local_ssrc: > >> ${CHANNEL(rtpqos,audio,local_ssrc)}) > > > > > > Are those ${CHANNEL( (rtpqos,audio,local_ssrc)} values available today in > an > > Asterisk version > > Yes. 1.4 and 1.6. But only for SIP channels obviously. > chan_sip.c: acf_channel_read()Thanks !> > > > Philipp Kempgen > > -- > http://www.das-asterisk-buch.de - http://www.the-asterisk-book.com > Amooma GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de > Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 > -- > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080918/26b2c713/attachment.htm