VoIP Mailing List Archives
Mailing list archives for the VoIP community |
|
View previous topic :: View next topic |
Author |
Message |
mninama at varaha.com Guest
|
Posted: Fri Jun 27, 2008 3:35 am Post subject: [asterisk-users] Asterisk cuts off intial voice path on brid |
|
|
I am using asterisk-1.4.21 and it is configured to pass media through it for
SIP calls. I have observed that if the callee answers the call and starts
speaking immediately for e.g. 'Hello one two three', the caller would get to
hear only 'one two three'. From packet captures I can see that asterisk
receives all the RTPs from the callee but it truncates the 'Hello' word from
the voice path when passing the stream on the other side.
The signaling gets complete between caller and callee, so asterisk should
bridge the channels immediately. I am using canreinvite=no and nat=yes
option in sip.conf.
Has anyone observed this issue why asterisk is cutting of the initial voice?
---Mayur
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080627/2888f249/attachment.htm |
|
Back to top |
|
|
oej at edvina.net Guest
|
Posted: Fri Jun 27, 2008 4:00 am Post subject: [asterisk-users] Asterisk cuts off intial voice path on brid |
|
|
27 jun 2008 kl. 10.35 skrev Mayur:
Quote: | I am using asterisk-1.4.21 and it is configured to pass media
through it for SIP calls. I have observed that if the callee answers
the call and starts speaking immediately for e.g. ?Hello one two
three?, the caller would get to hear only ?one two three?. From
packet captures I can see that asterisk receives all the RTPs from
the callee but it truncates the ?Hello? word from the voice path
when passing the stream on the other side.
The signaling gets complete between caller and callee, so asterisk
should bridge the channels immediately. I am using canreinvite=no
and nat=yes option in sip.conf.
Has anyone observed this issue why asterisk is cutting of the
initial voice?
|
I haven't observed it like this, but I now that when we send audio
over NAT, it takes a while to set up all media channels. We need to
receive RTP from both phones, in order to get a hole through the NAT
and be able to send audio out. That usually means that some RTP
packets we send before this happens is lost. Make sure you have turned
off silence suppression in both telephones, so that phones has no
delay in sending audio, even if it's just silence.
Regards,
/O |
|
Back to top |
|
|
|
|
|
You cannot post new topics in this forum You cannot reply to topics in this forum You cannot edit your posts in this forum You cannot delete your posts in this forum You cannot vote in polls in this forum
|
Powered by phpBB © 2001, 2005 phpBB Group
|