Hello everyone,
I have been having an issue where my SIP calls to another asterisk box (LAN) takes around 40 seconds before the call is crystal clear. Until then, audio will randomly cut out, you will hear things said 3-4 seconds ago. And then magically it all clears up and the call is perfect. Could anyone tell me what could be causing this strange issue? Also I have noticed “Probation passed - setting RTP source address to xx.x.x.x” on both SIP boxes does the call becomes crystal clear, is there a way to speed this up?
sip.conf
[general]
context=incoming-context
allowoverlap=yes
bindport=5060
bindaddr=0.0.0.0
srvlookup=no
qualify=5000
alwaysauthreject=yes
nat=yes
canreinvite=no
directrtpsetup=no
icesupport=yes
rfc2833compensate=yes
relaxdtmf=yes
dtmfmode=inband
allow=gsm,ulaw
This is an RTP effect, not a SIP one, and is generally the result of poor recovery by a jitter buffer when there is a sudden jump in timestamp.
Earlier versions of Asterisk failed to pass through the source identity (SSRC), but did pass through timestamps. They set the mark on the packet when there was a change, but the semantics of that are not the same.
I’m not sure that it is still true that Asterisk doesn’t pass on SSRCs, but it is definitely true that the ability to recover from such steps is very dependent on the recipient s system.
Thanks, that’s very informative. I suspected it to be RTP, but did not understand it’s purpose. I tried messing with rtp.conf, but i was just blindly guessing different values. Are there specific parameters that can make a significant difference you may know of and their recommended values? I suppose sip.conf’s jitter parameters may also play a role. Oh, I am using Asterisk 13.9. Thanks a lot