[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6
Chris Stone
axisml at gmail.com
Tue Feb 8 20:51:51 CET 2011
Dave,
On Tue, Feb 8, 2011 at 12:02 AM, Dave Singer <dave.singer at wideideas.com> wrote:
> Don't know what tools you are familiar with so here are some
> suggestions for what they're worth.
Appreciate the input!
Am familiar with all - but included output below - always happy to
have another set of eyes on things ;-)
> what is listening on port 5060?
> netstat -lnp | grep 5060
tcp 0 0 67.212.153.178:5060 0.0.0.0:*
LISTEN 25098/opensips
tcp 0 0 127.0.0.1:5060 0.0.0.0:*
LISTEN 25098/opensips
udp 0 0 67.212.153.178:5060 0.0.0.0:*
25098/opensips
udp 0 0 127.0.0.1:5060 0.0.0.0:*
25098/opensips
and as seen below, that's the only opensips pid on the system...
> is opensips actually running? what was the command line used?
> ps aux --forest | grep opensips
root 25098 0.0 0.0 72884 992 ? S 11:40 0:00
/usr/sbin/opensips
root 25100 0.0 0.0 72884 468 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25101 0.0 0.0 72884 468 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25102 0.0 0.0 72884 468 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25103 0.0 0.0 72884 468 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25104 0.0 0.0 72884 468 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25105 0.0 0.0 72884 472 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25106 0.0 0.0 72884 472 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25107 0.0 0.0 72884 472 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25108 0.0 0.0 72884 472 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25109 0.0 0.0 72884 472 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25110 0.0 0.0 72884 592 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25111 0.0 0.0 72884 600 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25112 0.0 0.0 72884 708 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25113 0.0 0.0 72884 708 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25114 0.0 0.0 72884 708 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25115 0.0 0.0 72884 708 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25116 0.0 0.0 72884 708 ? S 11:40 0:00 \_
/usr/sbin/opensips
root 25117 0.0 0.0 72884 704 ? S 11:40 0:00 \_
/usr/sbin/opensips
> Does sip still pass when opensips is not running?
Nope - inbound calls hang and then get a fast busy. Start opensips and
they ring through.
> if mi_fifo loaded, what is the output of <opensips install
> path>/sbin/opensipsctl fifo ps
> does it show that it is listening on the interfaces/ports that are
> handeling the packets you are capturing?
Wasn't loaded. Added it and retested call and yes, it shows listening
on the correct interfaces and ports - all interfaces and tcp/udp port
5060:
Process:: ID=0 PID=25256 Type=attendant
Process:: ID=1 PID=25257 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=2 PID=25259 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=3 PID=25260 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=4 PID=25261 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=5 PID=25262 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=6 PID=25263 Type=SIP receiver udp:67.212.153.178:5060
Process:: ID=7 PID=25264 Type=SIP receiver udp:67.212.153.178:5060
Process:: ID=8 PID=25265 Type=SIP receiver udp:67.212.153.178:5060
Process:: ID=9 PID=25266 Type=SIP receiver udp:67.212.153.178:5060
Process:: ID=10 PID=25267 Type=SIP receiver udp:67.212.153.178:5060
Process:: ID=11 PID=25268 Type=time_keeper
Process:: ID=12 PID=25269 Type=timer
Process:: ID=13 PID=25270 Type=MI FIFO
Process:: ID=14 PID=25271 Type=TCP receiver
Process:: ID=15 PID=25272 Type=TCP receiver
Process:: ID=16 PID=25273 Type=TCP receiver
Process:: ID=17 PID=25274 Type=TCP receiver
Process:: ID=18 PID=25275 Type=TCP receiver
Process:: ID=19 PID=25276 Type=TCP main
> If something else is listening on port 5060 opensips should be failing
> to start with the provided config.
Right, not failing to start though - no port conflict....
> try running single mode with the following config:
> and run it with:
> /usr/sbin/opensips -f <config file path/opensips.cfg> 2>&1 | tee test.log
> (path to opensips executable based on your mpath in the config.
>
> The debug went in to test.log as well as to the screen so you can look
> in it and search for the "test test" to see if it handled the
> message. Or details of why it failed to start.
Had to add a 'listen.....' line to the cfg file to get it to listen on
the public IP instead of only localhost. Started up and yes, the 'test
test' text showed up and the call exhibited the same behavior and does
still seem to change the IP in the SIP body (reason for the SDP being
routed via OpenSIPS??) and the Contact header too (someone else
noticed that - don't know how significant that is....).
Incoming INVITE to Opensips 1.6 server from upstream provider:
INVITE sip:17204497101 at 67.212.153.178:5060;transport=udp SIP/2.0\r\n
From: "STONE C AND C"
<sip:+13038382386 at 208.94.157.10:5060>;tag=a9d5ed0-13c4-4d519b73-1fad4e58-4134318e\r\n
To: <sip:17204497101 at 67.212.153.178:5060>\r\n
Call-ID: CXC-219-728c1330-a9d5ed0-13c4-4d519b73-1fad4e58-222f4090 at 208.94.157.10\r\n
CSeq: 1 INVITE\r\n
Via: SIP/2.0/UDP
208.94.157.10:5060;branch=z9hG4bK-1522a-4d519b73-1fad4e58-c268daf\r\n
Max-Forwards: 69\r\n
P-Asserted-Identity: "STONE C AND C "
<sip:+13038382386 at cxc.dashcs.com:5060>\r\n
Supported: timer,100rel\r\n
Content-Disposition: session;handling=required\r\n
Contact: <sip:+13038382386 at 208.94.157.10:5060;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 24600 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 INVITE from Opensips 1.6 server to Asterisk server:
INVITE sip:17204497101 at 67.212.153.178:5060;transport=udp SIP/2.0\r\n
From: "STONE C AND C"
<sip:+13038382386 at 208.94.157.10:5060>;tag=a9d5ed0-13c4-4d519b73-1fad4e58-4134318e\r\n
To: <sip:17204497101 at 67.212.153.178:5060>\r\n
Call-ID: CXC-219-728c1330-a9d5ed0-13c4-4d519b73-1fad4e58-222f4090 at 208.94.157.10\r\n
CSeq: 1 INVITE\r\n
Via: SIP/2.0/UDP
67.212.153.178:5060;branch=z9hG4bK-1522a-4d519b73-1fad4e58-c268daf\r\n
Via: SIP/2.0/UDP
208.94.157.10:5060;branch=z9hG4bK-1522a-4d519b73-1fad4e58-c268daf\r\n
Max-Forwards: 69\r\n
P-Asserted-Identity: "STONE C AND C "
<sip:+13038382386 at cxc.dashcs.com:5060>\r\n
Supported: timer,100rel\r\n
Content-Disposition: session;handling=required\r\n
Contact: <sip:+13038382386 at 67.212.153.178:5060;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 24600 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
And, just to confirm, this was with no parameters on the opensips
command line and with the config:
#-----------------------------------------------------------------------
debug=9 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
children=10
check_via=no # (cmd. line: -v)
dns=off # (cmd. line: -r)
rev_dns=off # (cmd. line: -R)
port=5060
# ------------------ module loading ----------------------------------
mpath="/usr/lib64/opensips/modules"
#loadmodule "xlog.so" # un comment if using opensips before 1.6.4
# ----------------- setting module-specific parameters ---------------
route{
# xlog("\n\n\n test test \n\n\n");
forward("67.212.153.179");
exit;
}
Very confusing - see nothing else on the server that would be touching
the packets - particularly while Opensips is processing them.......
Regards,
Chris
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