asterisk users - Jun 2006 - SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com

Dear list (and more specifically Bret),

I am getting one-way (inbound only) audio when trying to place a SIP call
via voip.trxtel.com (i.e. 18005558355@voip.trxtel.com). The Cli spits out
"== Forcing Marker bit, because SSRC has changed" 5 times after
atempting a
native bridge. I realize this is most certainly a NAT issue, the * server is
behind one. Sip.conf has externip=, and localnet=.

Can someone explain what SSRC changing implies is going on, and why it
affects NAT? This is actually the first NAT issue I have had with *, all
other SIP calls have worked fine (I must be lucky).


Sincerely,

Brent A. Torrenga

Torrenga Engineering, Inc.
907 Ridge Road
Munster, Indiana 46321-1771

tel:+1 219 836 8918 x325
fax:+1 219 836 1138
email:brent.torrenga@torrenga.com
web:www.torrenga.com

----- Brent Torrenga <lists@torrenga.com> wrote:
> "== Forcing Marker bit, because SSRC has changed" 5 times after
> atempting a
> native bridge. I realize this is most certainly a NAT issue, the *
> server is
> behind one. Sip.conf has externip=, and localnet=.
google is your friend :-) We've already covered this on the list, it's
not something that you should be concerned about (or even seeing). Later
releases of Asterisk no longer generate this message, and when they do it is not
marked as a warning, because it is not.

-- 
Kevin P. Fleming
Senior Software Engineer
Digium, Inc.