[Asterisk-Users] SIP One-way audio: == Forcing Marker bit, because SSRC has changed - trxtel.com

[Brent Torrenga]

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 at 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 at torrenga.com
web:www.torrenga.com