[Asterisk-Users] SIP and audio delay

[Jeremy Hall]

Kubat, Philip <> scribbled on Wednesday, June 23, 2004 2:32 PM:

> I have a SIP connection to Broadvoice and sometimes when I make
> outgoing calls from a SIP ATA-188 (could be the same number) (the
> ATA-188, is currently the only extension), there is no audio passed
> for 5-10 secs.  I have set all the codec the same to 711u and also
> ensured canreinvite is set to no.    
> 
> 
> 
> Any suggestions?  Places to look for?

Philip,

I'm not quite sure what you are asking here.  Are you connecting
directly to Broadvoice with your ATA-188?  If so, you would need to
contact Broadvoice support in this case.

If you are connecting to Broadvoice with Asterisk, then to Asterisk with
your ATA-188, you probably have either a configuration problem with your
.conf files or a firewall issue that is taking a long time to sort out.
Take a look at the Wiki page Asterisk Settings Broadvoice.
(http://www.voip-info.org/tiki-index.php?page=Asterisk%20settings%20Broa
dvoice)  It is based off of my configuration which gives me no delay at
all.  Since I am behind a NAT, I also forwarded UDP ports 5060 and the
port range listed in rtp.conf to my Asterisk box as well.  

Something else that may be causing you grief is the dial plan in your
ATA-188.  I am not familiar with that model as I am currently using a
Sipura, but you may have a case where you are dialing, and hitting the
digit timeout before the SIP initiation takes place.  Try dialing your
number then pressing the Pound (#) key and see if the dialing is
immediate.  If it is, check the configuration of your ATA and see if you
can either shorten the timeout period or create a pattern match that
will pick up the shorter dial string.

Here is what I would try (in this order) in your shoes:

1.  Try dialing with a # key and see if that fixes it.  Modify ATA
config if it does.
2.  Modify your firewall rules to forward the above mentioned ports to
your Asterisk box.
3.  Compare what you have in your sip.conf with the Wiki.  If you have
different configuration lines, modify yours to more closely match the
working example.
4.  If none of the above ideas work, reply with the relevant sections of
your sip.conf and extensions.conf to this thread and somebody may be
able to see something that is not correct.

Good luck,

Jeremy