[Asterisk-Users] Strange RTP audio errors on console

[Chris A. Icide]

I have a system running CVS HEAD 6/30/2004.  We've only been using it for 
PSTN to channel bank handsets, but have decided to add sip phones into the 
mix.  Now I have quite a few systems running sip phones just fine as well 
as some running both sip and analog via channel banks or tdm cards.

When we tried to set up some sip extensions (they are behind nats, we are 
using xten light, and have canreinvite=no as well as nat=yes set in the 
sip.conf), we only get one way audio.  You can hear the other end (be it 
the asterisk voice prompts or another non-sip user), but the other user 
cannot hear the sip phone user talking.

It gets even more complex.  If using the sip phone to call voicemail, or 
any other asterisk based services the sip user can get dtmf through (yes 
rfc2833).

The asterisk box is on a public IP address, no firewall or nat.  The sip 
clients are 'generally' behind a nat (we've tested from several locations 
including my home, which I have multiple sip UA's behind a nat, and the 
very same xten lite is able to work just fine with any of my local asterisk 
systems (or any others I've tested - all outside of my home nat).

The calls are being made with ulaw as the only codec allowed.  The sip 
debug indicates that the call setup has worked and agreed upon ulaw as the 
codec.

CLI provides multiple repeats of the following two errors:

rtp.c:1215 ast_rtp_write: Not sure about sending format SLINR packets
rtp.c:1058 ast_rtp_raw_write: Not sure about timestamp format for codec


Any thoughts, comments, suggestions?

Google has been less than helpful given any keywords I came up with to 
avoice all the NAT/Firewall one way audio posts.


-Chris