Olle has posted a writeup on his RTCP work - still could do with more testers:
Friends,
During my adventures in RTCP I've discovered that it's not only Asterisk that has been paying too little attention to RTCP...
Grandstream GXV3140 doesn't send any RTCP reports during a five minute call. And it's a video phone. Video really needs RTCP. I *hope* there's a setting somewhere that I've missed.
Polycom Soundpoint IP600 - propably with old firmware - sends NTP timestamps that Wireshark tells me are dated Feb 7, 2036. This means that we have many years of roundtrip time, measured in millisecs. For a phone connected to the same switch as my Asterisk server, that's a huge latency. I guess they need a bit more of CPU power for that device. :-) (or it's me that needs help to upgrade my phone to a version where they've fixed this bug).
SNOM 370 sends a timestamp that's all zeroes, so we can't measure RTT.
Cisco/Linksys sends a valid timestamp and Asterisk measures RTT properly.
Many phones only send RTCP at regular intervals. If this interval is longer than the call, we'll get no report at all from the other side and can't get any idea about QoS for the outbound stream. It should really be a requirement to send a final report at hangup time for every call.
RTCP is also affected by NAT, the same way as RTP and SIP. So if there's too much time between each RTCP report, NAT relationships will be forgotten and RTCP packets lost. There should be a setting for the RTCP timer, so you can make sure that NAT is kept open. Asterisk has that and in my "pinefrog-1.4" version also sends a packet to open up the NAT when media starts.
We need to get the industry to shape up here and treat RTCP seriously. It will become even more important with new codecs that use the stats from RTCP to adapt to current network conditions. If you buy a lot of phones from a vendor, require proper RTCP support in the bid. I'll try to update a blacklist document in my svn branch. So far, CounterPath Eyebeam is the winner - final report sent together with the GoodBye packet. Cisco SPA is the second on the list, even though I get no final report from that phone.
Please keep testing my test branch "pinefrog-deluxe-rtcp-test" and provide me with feedback!
RTCP In Devices Is Doing Better Than You Might Think Written by Neill Wilkinson - http://aeonvista.blogspot.com/ (May 24, 2010, 1:27 pm)
Olle,
Its nice to see someone on the case of RTCP between devices. I've been, for the last year, championing the use of RTCP-XR (RFC3611) which has a much more useful feature capability to include significantly more information about voice and video QoS.
The good news is both Snom and Polycom support RTCP-XR. Although for Snom this is very new and in the 8.2.x and 8.4.x branches of firmware. I have in my lab working versions of this code based on an Snom 360 and 320 phone. Polycom have been supporting RTCP-XR for some time and use an embedded agent from Telchemy to provide MOS scores as part of the RTCP-XR data.
Both Snom and Polycom are also compliant with http://tools.ietf.org/html/draft-ietf-sipping-rtcp-summary-10 RTCP-XR summary reporting in SIP PUBLISH messages. This means you can "collect" Call Quality data in a network (I don't want to use the over used phrase cloud) based reporting engine. This has the advantage of not having to be in the RTCP "stream" and the end devices can monitor the bidirectional RTCP-XR data and provide "near & far-end" QoS information to a central place.
Asterisk@home festival weather report - February 25, 2005 Dean Collins has posted a script to allow you to get a weather report via the Festival Speech Synthesis engine.
AstriDevCon: October 29th, Washington DC August 23, 2010 Average Vote: 10
John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.
VoIP-Info: FFasterisk Video file converter August 25, 2006 Average Vote: 10
The wiki has a link to a new piece of software for converting video to the format required for Asterisk.
Voip-Forum: Lots of new articles March 12, 2005 Average Vote: 10
Oej's Voip-Forum.com site has posted lots of new news articles while I've been away. Hopefully you found them via the asterisk-docs site. If not I've bookmarked them for you.
Interview with Mark Spencer November 26, 2004 Average Vote: 9.9
We have managed to get an interview with Mark Spencer AKA Markster. Mark Spencer is the creator of Asterisk and by far the most active developer.
Back to life July 21, 2010 Average Vote: 9.8
Hey all - I am back online after some pretty big projects which have taken all my time. Will be updating the Asterisk news over the next few days.
GUI changes from Trixbox, FreePBX, 2600hz, BlueBox September 1, 2010 Average Vote: 9.8
Ok, bear with me on this one. If you understand all the ramifications, FreePBX has split to a new project called BlueBox contained within the 2600hz project. This obviously has implications for Trixbox that uses FreePBX to provide quite a bit of functionality.
Announcing Adhearsion 0.8.5 August 25, 2010 Average Vote: 9.8
Ben Klang has posted a note about the latest release of Adhearsion - a framework for developing Asterisk based solutions using Ruby.
app_swift v2.0 released July 21, 2010 Average Vote: 9.8
Like a few of these news stories that I will be posting over the next couple of days this is a little old - hope it is not something you have already seen. This one is for a new version of the app_swift text-to-speech module for Asterisk 1.2, 1.4, and 1.6.
Monitoring Asterisk with Munin January 7, 2010 Average Vote: 9.7
I had a few requests for these munin plugins after some discussion on one of the Asterisk lists and thought people might like them.
libpri 1.4.11.4 Now Available September 3, 2010 The Asterisk Development Team has announced the release of libpri 1.4.11.4.
New CDR Stats Package September 1, 2010 This one has been a long time coming. A new CDR stats package from Star2Billing to replace the 7 year old stalwart for viewing Asterisk call detail records.
GUI changes from Trixbox, FreePBX, 2600hz, BlueBox September 1, 2010 Ok, bear with me on this one. If you understand all the ramifications, FreePBX has split to a new project called BlueBox contained within the 2600hz project. This obviously has implications for Trixbox that uses FreePBX to provide quite a bit of functionality.
RazorQuotePBP Asterisk Payment Module August 31, 2010 RazorQuote has sent us a press release about the launch of RazorQuotePBP, a native Asterisk module that allows any Asterisk connected device to accept credit card payments.
AstriCon approaches August 25, 2010 John Todd has posted a note about the upcoming AstriCon conference in Washington, DC, and the innovation awards.
Announcing Adhearsion 0.8.5 August 25, 2010 Ben Klang has posted a note about the latest release of Adhearsion - a framework for developing Asterisk based solutions using Ruby.
AstriDevCon: October 29th, Washington DC August 23, 2010 John Todd has posted a note about the AstriDevCon conference which occurs within the Astricon conference.
The XV Commandments of IVR August 17, 2010 An update on the 15 tips for creating effective IVR systems by Allison Smith - the Voice of Asterisk.
RTCP In Devices Is Doing Better Than You Might Think