Asterisk 10.4.0-rc1
At outgoing call from sip to dahdi(pri) i see a lot these messages :
[2012-04-26 11:06:57] DEBUG[30214] res_rtp_asterisk.c: Setting the marker bit due to a source update
They disappear after connect. What is wrong ?
Nothing. Or rather, you have your debug level set too high. Debug messages do not indicate a problem, although they may help diagnose one.
RTP packets have a synchronisation source field. This should change when the source of the timestamps changes, although the standard version of Asterisk doesn’t do this, unless it is native bridging (in which case it just passes it through). They also have a marker bit, whose semantics are a bit fuzzy, but one use of which is to indicate points in the audio stream where latency buffer adjustments can safely be made.
Even when not native bridging, Asterisk treats synchonisation source changes as candidate point of latency buffer resets.
(Note that we have found that Cisco phones react badly if the timestamps have discontinuities, but the synchronisation source doesn’t change, so we added a local change, but only in an unsupported version, to cause the outgoing synchronisation source to change whenever a frame with a marker bit is output.)