|Summary:||ASTERISK-07281: RTP Packets go through Asterisk|
|Reporter:||Ken Chan (ck000)||Labels:|
|Date Opened:||2006-07-04 13:41:51||Date Closed:||2006-08-02 15:26:12|
|Environment:||Attachments:||( 0) sip_log_for_7479bug_2.txt|
( 1) sip_log_for_7479bug_3.txt
( 2) sip_log_for_7479bug.txt
|Description:||I am using Asterisk-Addon ooh323 channel driver (1.2.3) and Asterisk 184.108.40.206.|
Establish a call between SIP and OpenPhone. The RTP Packets are going from
OpenPhone --> Asterisk --> SIP phone.
Both channel drivers (ooh323 and sip) use "ast_rtp_bridge" and the RTP streams shall go from endpoint to endpoint
****** ADDITIONAL INFORMATION ******
In CLI, I used "rtp debug ip 10.3.2.111" to confirm that the RTP packets went through Asterisk.
|Comments:||By: Serge Vecher (serge-v) 2006-07-05 10:57:40|
1) Prepare test environment (reduce the ammount of unrelated traffic on the server);
2) Make sure your logger.conf has the following line:
console => notice,warning,error,debug
3) restart Asterik.
4) Enable SIP transaction logging with the following CLI commands:
set debug 4
set verbose 4
5) Save complete log to file and _attach_ said file to the bug.
By: Serge Vecher (serge-v) 2006-07-05 11:00:29
this was recently discussed on the asterisk-dev mailing list. http://lists.digium.com/pipermail/asterisk-dev/2006-June/021396.html
By: Ken Chan (ck000) 2006-07-10 13:14:38
Sorry for the late reply. Busy with other stuff.
Attached please found the capture info you requested for.
Here are my setup:
Asterisk -- 10.3.3.239
SIP phone ("voip6111" -- Extension is 6111) -- 10.3.2.111
H323 Open Phone ("ken_op" -- Extension is 7401) -- 10.1.1.155
Ignore all the messages generated by IP "10.1.1.160" or "x-lite-john-pc".
Basically, SIP phone dialed "7401" to connect to H323 Open Phone. Then the RTP packets from Open Phone went through Asterisk to SIP Phone. (In CLI>, I typed "rtp debug" to confirm).
If you need more log, please do let me know.
By: Serge Vecher (serge-v) 2006-07-10 14:21:00
interesting ... Asterisk sends a REINVITE back 6111 and out of the blue we get a hangup.
Jul 10 17:50:04 DEBUG: rtp.c:1801 ast_rtp_bridge: Oooh, got a hangup
Unfortunately, this doesn't show who initiated it, but looks like it was chan_ooh323.
I did notice that you use an agi script for dialing. Can you please test with the Dial() command instead? While you are redoing the test, maybe you could eliminate the pesky x-lite out of the logs too. Thanks.
By: Ken Chan (ck000) 2006-07-10 16:22:08
Thanks for the quick reply.
In my 1st attachement, I think I disconnected the call and that was why you saw the hang-up message. sorry.
Please see the 2nd attachement.
I changed my extension.conf to "exten => 7401,1,Dial(OOH323/ken_op,20)".
I did not disconnect the call this time. Hopefully I provide enough info this time.
Do you want to see my ooh323.conf? Or turn on h323 debug?
By: Serge Vecher (serge-v) 2006-07-10 16:31:08
Yes, might as well see what the ooh323 channel is doing. Let see what it looks like with both debugs on.
By: Ken Chan (ck000) 2006-07-10 17:01:16
The 3rd attachment contains debug messages for both SIP and OOH323.
By: Ken Chan (ck000) 2006-07-13 14:53:37
What is the status of this one?
By: Kevin P. Fleming (kpfleming) 2006-08-02 15:26:11
According to many H.323 developers I have spoken with, there is no way to do a re-invite of the media stream in H.323, so what you want to accomplish is not possible.