[asterisk-commits] oej: branch oej/pinefrog-rtcp-1.8 r383164 - /team/oej/pinefrog-rtcp-1.8/

[SVN commits to the Asterisk project]

Author: oej
Date: Fri Mar 15 05:57:38 2013
New Revision: 383164

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=383164
Log:
Adding some explanations on why you will not get RTCP statistics for all calls.
Please put some pressure on your vendors here.

Modified:
    team/oej/pinefrog-rtcp-1.8/README.pinefrog-rtcp

Modified: team/oej/pinefrog-rtcp-1.8/README.pinefrog-rtcp
URL: http://svnview.digium.com/svn/asterisk/team/oej/pinefrog-rtcp-1.8/README.pinefrog-rtcp?view=diff&rev=383164&r1=383163&r2=383164
==============================================================================
--- team/oej/pinefrog-rtcp-1.8/README.pinefrog-rtcp (original)
+++ team/oej/pinefrog-rtcp-1.8/README.pinefrog-rtcp Fri Mar 15 05:57:38 2013
@@ -61,6 +61,26 @@
 - It doesn't handle re-invites in a good way.
 - It seems to mix sender and receiver reports, thus mixing data from two streams 
     - when does this happen, if at all?
+
+NOTES
+-----
+RTCP is a mutual protocol. Asterisk sends data to a phone and tells the phone in a
+"Sender report" how much we've sent. The phone responds with a "Receiver report"
+to give data about packets lost between the sender and receiver, variations in
+time (jitter) and timestamps to calculate latency. The phone does the same
+with the RTP stream towards Asterisk - sends a "Sender Report" and receives
+a "Receiver report". 
+
+This means that in order to get relevant data, it's a tango for two. The phone
+needs to have a working RTCP implementation. This is sadly not the case for 
+all SIP phones. Some phones have timers, so for short calls you will not get
+any reports, because it's set to send RTCP after five minutes or something
+longer than your phone call. 
+
+The conclusion is that the CQRs doesn't work for all phones. You will always
+get data from Asterisk's point of view, but you may not get data from the other
+end. Put pressure on your vendors to participate in the RTCP interaction
+so you can get control of your calls.
 
 RTCP and NAT
 ------------