[asterisk-users] SIP Delay with remote stations?
Benny Amorsen
benny+usenet at amorsen.dk
Thu Jul 1 04:29:08 CDT 2010
"William Stillwell (Lists)" writes:
> I have several remote phones that experience a slight call delay when
> answering phones, ie, they will answer, speak a few words, and then the
> remote caller will hear them, and the first half is cutoff?
This is actually a somewhat common problem in SIP. One end sends media
before the other end is ready to receive it, or a gateway receives media
on one leg of the call but media isn't yet ready on the other leg...
In your case I would guess that it is caused by firewalls/NAT reacting
only to RTP traffic in one direction, thereby blocking traffic in the
other until the first packet.
Luckily it's IP, so you can use tcpdump or wireshark or phone-specific
dump tools to capture the traffic and see where the problem hides.
/Benny
More information about the asterisk-users
mailing list