Summary:	ASTERISK-20758: No audio when P2P bridge occurs
Reporter:	TSAREGORODTSEV Yury (tsarik)	Labels:
Date Opened:	2012-11-30 12:11:34.000-0600	Date Closed:	2013-01-31 10:12:32.000-0600
Priority:	Critical	Regression?
Status:	Closed/Complete	Components:	Core/RTP
Versions:	1.8.18.0 11.0.1	Frequency of Occurrence
Related Issues:
Environment:		Attachments:
Description:	sip.conf [general] directmedia=no directrtpsetup=no [peer1] directmedia=no directrtpsetup=no canreinvite=no [peer2] directmedia=no directrtpsetup=no canreinvite=no But anyway * Send P2P packets. Only in case of force jitter buffer * bridging channels locally. But to use jitterbuffer not a right solution in this case.
Comments:	By: Joshua C. Colp (jcolp) 2012-11-30 12:13:34.862-0600 Unless there is some sort of other issue here this is expected behavior. P2P bridging is an efficient packet forwarding bridging done within the RTP stack. Media still flows through Asterisk. By: TSAREGORODTSEV Yury (tsarik) 2012-11-30 12:16:11.661-0600 WHo said media still flow through Asterisk? Not at all just P2P packets send and thats its. No RTP packets go in that case, people cant hear each other. By: Joshua C. Colp (jcolp) 2012-11-30 12:20:29.609-0600 That means media is still flowing through Asterisk. It receives the RTP packet, modifies it minimally, and sends it back out. If you are having issues with people not hearing each other I would suggest you look into NAT issues and ask on the asterisk-users list for help. The problem is not with P2P bridging. By: TSAREGORODTSEV Yury (tsarik) 2012-11-30 12:29:23.697-0600 There is no NAT issue, noone under NAT. Why asterisk modify packets if directmedia=no ? Why thats never happened while jbforce=yes ? When jbforce=yes asterisk never send P2P and users can talk. By: Joshua C. Colp (jcolp) 2012-11-30 12:32:21.170-0600 When media is not going directly Asterisk always deconstructs the RTP packets, because the other side may not be an RTP channel. In the case of a P2P bridge since both sides are RTP the packet is minimally modified to make sure that payload numbers and such are correct. It is incredibly strange that with a jitterbuffer forced the users can talk, since implementing the P2P bridge over 5 years ago this is the first time I've ever heard of that. This may be something to do with the specific phones on each side. By: TSAREGORODTSEV Yury (tsarik) 2012-11-30 12:41:54.351-0600 Dear Joshua, question is why while directmedia=no asterisk still sent P2P packets and in case jbforce=no asterisk never send them. How to turn off p2p packets without usage of jitterbuffer. By: Joshua C. Colp (jcolp) 2012-11-30 12:46:24.344-0600 P2P bridging is *not* the same as directmedia. Direct media has the devices communicate directly and not through Asterisk. P2P bridging occurs within Asterisk and should just work when conditions allow it to. As I said previously this is the first time in many years that it hasn't just worked for someone, so something unique to you is probably causing issues. By: TSAREGORODTSEV Yury (tsarik) 2012-11-30 12:48:25.980-0600 Is there any solution to turn off P2P? By: Joshua C. Colp (jcolp) 2012-11-30 12:49:28.709-0600 Besides doing as you have done and forcing the jitterbuffer or modifying the code, no. Nobody should ever have to turn it off, your case is unique By: TSAREGORODTSEV Yury (tsarik) 2012-11-30 12:51:34.244-0600 what kind of debug/log do you need to investigate my unique case? By: Joshua C. Colp (jcolp) 2012-11-30 12:53:49.307-0600 To look into this issue we'll need an "rtp debug" of working and non-working case, as well as a packet capture using wireshark or tcpdump of each. The model and firmware version of the devices on each side would also be useful, as well as the *exact* version of Asterisk. By: Rusty Newton (rnewton) 2013-01-04 15:06:15.573-0600 Is issue still occurring? Can you provide the debug requested? By: Rusty Newton (rnewton) 2013-01-31 10:12:32.751-0600 Suspending this until we get debug demonstrating an issue, as we have had no response for 2 weeks.  If you want to re-open, please also provide an asterisk full log with VERBOSE and DEBUG messages enabled at least at level 5. You can always contact Asterisk bug marshals in #asterisk-bugs if you have questions.