Hi Chris,
That config should't touch the Contact header, and yet that's also been
modified:
In: Contact:<sip:[email protected] ...
Out: Contact:<sip:[email protected] ...
Are you sure nothing else is touching the message?
Regards,
Henk Hesselink
On 08-02-11 02:33, Chris Stone wrote:
Ovidiu,
On Mon, Feb 7, 2011 at 4:19 PM, Ovidiu Sas<[email protected]> wrote:
By default, opensips does not modify the SDP.
Double check your config. If you don't need to touch SDP, make sure
that you are not loading nathelper or mediaproxy. Those are the two
modules that are changing SDP.
Made sure neither of these were being loaded and used - mediaproxy
was, but nathelper was not. I need neither, so removed, restarted
opensips, tested a call. No change - problem persisted. So, dropped
down to a bare config:
#-----------------------------------------------------------------------
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
children=25
check_via=no # (cmd. line: -v)
dns=off # (cmd. line: -r)
rev_dns=off # (cmd. line: -R)
port=5060
# for more info: sip_router -h
# ------------------ module loading ----------------------------------
mpath="/usr/lib64/opensips/modules"
# ----------------- setting module-specific parameters ---------------
route{
forward("67.212.153.179");
exit;
}
#-----------------------------------------------------------------------
Restarted OpenSIPS with the above, and problem persists - SDP routing
modified (apparently) and Opensips proxies the audio.
Incoming from upstream:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0\r\n
From: "STONE C AND C"
<sip:[email protected]:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n
To:<sip:[email protected]:5060>\r\n
Call-ID:
CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a@208.94.157.10\r\n
CSeq: 1 INVITE\r\n
Via: SIP/2.0/UDP
208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
Max-Forwards: 69\r\n
P-Asserted-Identity: "STONE C AND C "
<sip:[email protected]:5060>\r\n
Supported: timer,100rel\r\n
Content-Disposition: session;handling=required\r\n
Contact:<sip:[email protected]:5060;maddr=208.94.157.10;transport=udp>\r\n
Session-Expires: 1800\r\n
Content-Type: application/sdp\r\n
Content-Length: 238\r\n
\r\n
v=0\r\n
o=Acme_UAS 0 1 IN IP4 208.94.157.10\r\n
s=SIP Media Capabilities\r\n
c=IN IP4 208.94.157.10\r\n
t=0 0\r\n
m=audio 22684 RTP/AVP 0 18 101\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:18 G729/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=maxptime:20\r\n
a=sendrecv\r\n
Outgoing to Asterisk:
INVITE sip:[email protected]:5060;transport=udp SIP/2.0\r\n
From: "STONE C AND C"
<sip:[email protected]:5060>;tag=a9d5ed0-13c4-4d509b92-1bc5e644-648c7598\r\n
To:<sip:[email protected]:5060>\r\n
Call-ID:
CXC-260-758763d0-a9d5ed0-13c4-4d509b92-1bc5e643-43d2bf0a@208.94.157.10\r\n
CSeq: 1 INVITE\r\n
Via: SIP/2.0/UDP
67.212.153.178:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
Via: SIP/2.0/UDP
208.94.157.10:5060;branch=z9hG4bK-11150e-4d509b92-1bc5e644-273291f1\r\n
Max-Forwards: 69\r\n
P-Asserted-Identity: "STONE C AND C "
<sip:[email protected]:5060>\r\n
Supported: timer,100rel\r\n
Content-Disposition: session;handling=required\r\n
Contact:<sip:[email protected]:5060;maddr=208.94.157.10;transport=udp>\r\n
Session-Expires: 1800\r\n
Content-Type: application/sdp\r\n
Content-Length: 240\r\n
\r\n
v=0\r\n
o=Acme_UAS 0 1 IN IP4 67.212.153.178\r\n
s=SIP Media Capabilities\r\n
c=IN IP4 67.212.153.178\r\n
t=0 0\r\n
m=audio 22684 RTP/AVP 0 18 101\r\n
a=rtpmap:0 PCMU/8000\r\n
a=rtpmap:18 G729/8000\r\n
a=rtpmap:101 telephone-event/8000\r\n
a=maxptime:20\r\n
a=sendrecv\r\n
I've got to be missing something stupid - the setup works great under
1.4 - would expect as well or better under 1.6 - but appears that
there's some config option or default that I'm missing....
But, with such a basic config as above, not sure what it would
be.....Would sure seem that, by some default, OpenSIPS proxies the
audio, no?
Thanks!
Chris
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