Summary:	ASTERISK-02867: SIP to H323 Bridge Issue
Reporter:	nirsim (nirsim)	Labels:
Date Opened:	2004-11-22 07:14:05.000-0600	Date Closed:	2011-06-07 14:10:02
Priority:	Minor	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) sip_to_h323_no-bridge
Description:	When placing calls from an Asterisk SIP gateway connection to an H323 termination via Asterisk gateway (chan_h323), bridging of RTP isn't performed. ****** ADDITIONAL INFORMATION ****** The synopsis of the problem is as following: Technical Schematic: PSTN->E1 PRI->(*1)->SIP->(*2)->H323->GnuGK->Cisco (*1) - Asterisk Box 1 (*2) - Asterisk Box 2 When placing a call from a remotely connected PSTN Asterisk (*1) via SIP, to a second Asterisk SIP gateway (*2), which is then supposed to terminate the call via H323 (chan_h323) to a remote GnuGK configured in H245 routing mode, the call reaches the remote phone, however, no voice is heard. At first, I thought this may be a codec issue, or an H323 issue, however, when performing a similar test when (*1) is replaced with an X-Lite SIP client, test works fine. The sypmtom seen is as following: a. (*1) places SIP call to (*2) b. (*2) places H323 call to GnuGK c. The phone at the remote end is answered. d. Asterisk doesn't notice the answering and doesn't   bridge between SIP to H323.  Again, this situation happens only on an * to * SIP connect. I hadn't tried this using IAX2 yet, and am not entirely sure if that would help. however, I'll do a test soon to see.  Please find attached a debug/trace log from (*2), including the actual dialing.
Comments:	By: Brian West (bkw918) 2004-11-22 09:11:23.000-0600 1. Its not MAJOR.. read the bug posting guidelines. 2. Its not designed to brige the RTP.  Thus is not really a bug. By: nirsim (nirsim) 2004-11-22 17:57:21.000-0600 1. After reading the bug posting guidelines, and extracting from there:  "MAJOR: A bug which completely prevents Asterisk from operating in a method  that it normally is expected to operate -- and particularly if it cannot be  reasonably worked around -- is MAJOR. Significant protocol violations that  are not simply policy decisions are MAJOR."  While the above mentioned functionality I've described worked in previous  versions, I would assume that it is the NORMAL operation expected from  Asterisk. So, to my understanding, this is MAJOR. If I've reported it as  MAJOR, believe me I've read the guidelines for reporting. 2. I suppose the proper term I should have used was PROXYing, and not BRIDGing.   However, if I've actually been doing something I was not supposed to be   doing for the past year, please elaborate. By: Brian West (bkw918) 2004-11-22 18:53:33.000-0600 If I recall chan_h323 has always proxy'ed the rtp stream thru asterisk. bkw By: nirsim (nirsim) 2004-11-23 08:35:13.000-0600 Well, in that case, let me re-phrase the bug report: When placing calls from an Asterisk SIP gateway connection to an H323 termination via Asterisk gateway (chan_h323), proxying of RTP isn't performed. The synopsis of the problem is as following: Technical Schematic: PSTN->E1 PRI->(*1)->SIP->(*2)->H323->GnuGK->Cisco (*1) - Asterisk Box 1 (*2) - Asterisk Box 2 When placing a call from a remotely connected PSTN Asterisk (*1) via SIP, to a second Asterisk SIP gateway (*2), which is then supposed to terminate the call via H323 (chan_h323) to a remote GnuGK configured in H245 routing mode, the call reaches the remote phone, however, no voice is heard. At first, I thought this may be a codec issue, or an H323 issue, however, when performing a similar test when (*1) is replaced with an X-Lite SIP client, test works fine. The sypmtom seen is as following: a. (*1) places SIP call to (*2) b. (*2) places H323 call to GnuGK c. The phone at the remote end is answered. d. Asterisk doesn't notice the answering and doesn't    proxy the RTP between SIP to H323.   Again, this situation happens only on an * to * SIP connect. I hadn't tried this using IAX2 yet, and am not entirely sure if that would help. however, I'll do a test soon to see.   Please find attached a debug/trace log from (*2), including the actual dialing.  Now, a little information to be added to the issue is as following: I've tested an older box that I have, that uses and old CVS version of Asterisk and the old chan_h323 channel code. The respective CVS version is: CVS-HEAD-08/09/04-15:55:03, which is able to perform the RTP proxy from SIP to H323 without any visible problems. However, I'm fully aware that this specific version suffers from the Channel_Dead_Lock issue. So, I would assume that somewhere along the way, since 08/09 till today, something had been done to the RTP proxy functionality, that casued this to break. By: Olle Johansson (oej) 2004-12-12 16:01:05.000-0600 No activity, no patch. By: tdriscoll (tdriscoll) 2004-12-12 19:30:23.000-0600 Reminder sent to dimitel How is it that the issue with h323 to sip is closed? It is a problem for both the release and head version.