[OpenSIPS-Users] OpenSIPS-mediaproxy, Asterisk (packet2packet bridge) - problem with RTP

Wed Jul 1 13:33:15 CEST 2009

Hi,
Fist of all,  I would like to express my apologies for sending
multiple mails to the list.

> Firstly I would strongly suggest that you only send your question once,
> sending it more than once will not get it answered sooner.
>
> Secondly, I do not know what a Asterisk packet2packet bridge or a ring group
> is and have no intention of finding out.

Second, Asterisk Packet2Packet Bridging is => Audio is not going
through the Asterisk core,it comes into the RTP stack and goes
directly out. This decreases the amount of memory allocation that
happens, and things require less processing.

>
> Thirdly, I would suggest that you include logs from the relay from the last
> two scenarios you described.
>

Media Relay Logs

[TOPOLOGY]

SBC(Opensips + mediaproxy+LCR) --> A.B.C.204
Softswitch (ISDN)(A.B.C.210)
Voice Gateways ---> (A.B.C.246  A.B.C.58)
asterisk --->  (F.G.H.252)
-------------------------------------------------------------------------------------
End Topology ----------------------

Scenario A

debug: Received new SDP offer
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10000
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10001
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10002
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10003
debug: Added new stream: (audio) A.B.C.246:22946 (RTP: Unknown, RTCP:
Unknown) <-> A.B.C.204:10000 <-> A.B.C.204:10002 <-> Unknown (RTP:
Unknown, RTCP: Unknown)
debug: created new session 1b6bd467-2478eb86-21c09d4f-60cc at A.B.C.210:
3500 at A.B.C.210 (168961745) --> 6798 at A.B.C.204
debug: Received new SDP offer
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10004
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10005
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10006
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10007
debug: Added new stream: (audio) F.G.H.252:10680 (RTP: Unknown, RTCP:
Unknown) <-> A.B.C.204:10004 <-> A.B.C.204:10006 <-> Unknown (RTP:
Unknown, RTCP: Unknown)
debug: created new session 1e2fbb537249acc7482c94af59aaa11d at F.G.H.252:
3500 at F.G.H.252 (as001533d2) --> 7013 at A.B.C.204
debug: Got traffic information for stream: (audio) F.G.H.252:10680
(RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10004 <-> A.B.C.204:10006
<-> Unknown (RTP: A.B.C.246:28804, RTCP: Unknown)
debug: updating existing session
1e2fbb537249acc7482c94af59aaa11d at F.G.H.252: 3500 at F.G.H.252
(as001533d2) --> 7013 at A.B.C.204
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio)
F.G.H.252:10680 (RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10004 <->
A.B.C.204:10006 <-> A.B.C.246:28804 (RTP: A.B.C.246:28804, RTCP:
Unknown)
debug: updating existing session
1b6bd467-2478eb86-21c09d4f-60cc at A.B.C.210: 3500 at A.B.C.210 (168961745)
--> 6798 at A.B.C.204
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio)
A.B.C.246:22946 (RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) A.B.C.246:22946
(RTP: A.B.C.246:22946, RTCP: Unknown) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) F.G.H.252:10680
(RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10004 <-> A.B.C.204:10006
<-> A.B.C.246:28804 (RTP: A.B.C.246:28804, RTCP: A.B.C.246:28805)
debug: Got traffic information for stream: (audio) A.B.C.246:22946
(RTP: A.B.C.246:22946, RTCP: A.B.C.246:22947) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: Unknown, RTCP: Unknown)
debug: removing session 1e2fbb537249acc7482c94af59aaa11d at F.G.H.252:
3500 at F.G.H.252 (as001533d2) --> 7013 at A.B.C.204
debug: updating existing session
1b6bd467-2478eb86-21c09d4f-60cc at A.B.C.210: 3500 at A.B.C.210 (168961745)
--> 6798 at A.B.C.204
debug: Received updated SDP answer
debug: Unchanged stream: (audio) A.B.C.246:22946 (RTP:
A.B.C.246:22946, RTCP: A.B.C.246:22947) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: Unknown, RTCP: Unknown)
(Port 10004 Closed)
(Port 10005 Closed)
(Port 10006 Closed)
(Port 10007 Closed)
debug: Got traffic information for stream: (audio) A.B.C.246:22946
(RTP: A.B.C.246:22946, RTCP: A.B.C.246:22947) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: F.G.H.252:11378, RTCP:
Unknown)
debug: Got traffic information for stream: (audio) A.B.C.246:22946
(RTP: A.B.C.246:22946, RTCP: A.B.C.246:22947) <-> A.B.C.204:10000 <->
A.B.C.204:10002 <-> F.G.H.252:11378 (RTP: F.G.H.252:11378, RTCP:
F.G.H.252:11379)
debug: removing session 1b6bd467-2478eb86-21c09d4f-60cc at A.B.C.210:
3500 at A.B.C.210 (168961745) --> 6798 at A.B.C.204
(Port 10000 Closed)
(Port 10001 Closed)
(Port 10002 Closed)
(Port 10003 Closed)
<-------------------------------------------------------------------
END A scenario --------------------------------------------------------------------------------------------->

Scenario B (failed RTP connection-  Unknown IP of Asterisk)

debug: Received new SDP offer
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10008
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10009
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10010
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10011
debug: Added new stream: (audio) A.B.C.58:26890 (RTP: Unknown, RTCP:
Unknown) <-> A.B.C.204:10008 <-> A.B.C.204:10010 <-> Unknown (RTP:
Unknown, RTCP: Unknown)
debug: created new session 6e21089d-30080486-1055ce54-67fc at A.B.C.210:
3500 at A.B.C.210 (1208656243) --> 6798 at A.B.C.204
debug: Received new SDP offer
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10012
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10013
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10014
mediaproxy.mediacontrol.StreamListenerProtocol starting on 10015
debug: Added new stream: (audio) F.G.H.252:18138 (RTP: Unknown, RTCP:
Unknown) <-> A.B.C.204:10012 <-> A.B.C.204:10014 <-> Unknown (RTP:
Unknown, RTCP: Unknown)
debug: created new session 0e55309a5e5446a95f1160a227508a81 at F.G.H.252:
3500 at F.G.H.252 (as52fad00f) --> 7013 at A.B.C.204
debug: Got traffic information for stream: (audio) F.G.H.252:18138
(RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10012 <-> A.B.C.204:10014
<-> Unknown (RTP: A.B.C.246:24684, RTCP: Unknown)
debug: updating existing session
0e55309a5e5446a95f1160a227508a81 at F.G.H.252: 3500 at F.G.H.252
(as52fad00f) --> 7013 at A.B.C.204
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio)
F.G.H.252:18138 (RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10012 <->
A.B.C.204:10014 <-> A.B.C.246:24684 (RTP: A.B.C.246:24684, RTCP:
Unknown)
debug: updating existing session
6e21089d-30080486-1055ce54-67fc at A.B.C.210: 3500 at A.B.C.210 (1208656243)
--> 6798 at A.B.C.204
debug: Received updated SDP answer
debug: Got initial answer from callee for stream: (audio)
A.B.C.58:26890 (RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10008 <->
A.B.C.204:10010 <-> F.G.H.252:13836 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) A.B.C.58:26890
(RTP: A.B.C.58:26890, RTCP: Unknown) <-> A.B.C.204:10008 <->
A.B.C.204:10010 <-> F.G.H.252:13836 (RTP: Unknown, RTCP: Unknown)
debug: updating existing session
0e55309a5e5446a95f1160a227508a81 at F.G.H.252: 3500 at F.G.H.252
(as52fad00f) --> 7013 at A.B.C.204
debug: Received updated SDP answer
debug: Unchanged stream: (audio) F.G.H.252:18138 (RTP: Unknown, RTCP:
Unknown) <-> A.B.C.204:10012 <-> A.B.C.204:10014 <-> A.B.C.246:24684
(RTP: A.B.C.246:24684, RTCP: Unknown)
debug: updating existing session
6e21089d-30080486-1055ce54-67fc at A.B.C.210: 3500 at A.B.C.210 (1208656243)
--> 6798 at A.B.C.204
debug: Received updated SDP answer
debug: Unchanged stream: (audio) A.B.C.58:26890 (RTP: A.B.C.58:26890,
RTCP: Unknown) <-> A.B.C.204:10008 <-> A.B.C.204:10010 <->
F.G.H.252:13836 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) A.B.C.58:26890
(RTP: A.B.C.58:26890, RTCP: A.B.C.58:26891) <-> A.B.C.204:10008 <->
A.B.C.204:10010 <-> F.G.H.252:13836 (RTP: Unknown, RTCP: Unknown)
debug: Got traffic information for stream: (audio) F.G.H.252:18138
(RTP: Unknown, RTCP: Unknown) <-> A.B.C.204:10012 <-> A.B.C.204:10014
<-> A.B.C.246:24684 (RTP: A.B.C.246:24684, RTCP: A.B.C.246:24685)
debug: removing session 0e55309a5e5446a95f1160a227508a81 at F.G.H.252:
3500 at F.G.H.252 (as52fad00f) --> 7013 at A.B.C.204
(Port 10012 Closed)
(Port 10013 Closed)
(Port 10014 Closed)
(Port 10015 Closed)
debug: removing session 6e21089d-30080486-1055ce54-67fc at A.B.C.210:
3500 at A.B.C.210 (1208656243) --> 6798 at A.B.C.204
(Port 10008 Closed)
(Port 10009 Closed)
(Port 10010 Closed)
(Port 10011 Closed)

Regards,
Dimitris

On Tue, Jun 30, 2009 at 7:42 PM, Ruud Klaver<ruud at ag-projects.com> wrote:
> Hi,
>
> On 29 Jun 2009, at 08:14, Dimitrios Giannakopoulos wrote:
>
>> Hi,
>>
>> I have implemented the following scenario:
>>
>> [incoming pstn]--->[opensips]-->[asterisk] --->[sip phone]
>>                                                    |
>> [outgoing pstn]<---[opensips]<------|
>>
>> Opensips acts as SBC with mediaproxy functionality. Moreover, I use
>> the LCR module to route calls.
>> The Asterisk is located at the public domain and we have activated the
>> packet2packet bridge. A soft phone is registered to asterisk and we
>> have created a ring group that sends an incoming call to soft phone
>> and external line (outbound pstn) that rings simultaneous both
>> devices. Opesips version 1.4.5 or 1.5 Asterisk version 1.6
>>
>> Single calls without ring gourp:
>>
>> Incoming calls from PSTN to asterisk through Opensips with mediaproxy
>> enabled. It works properly.
>> Outgoing calls from Asterisk to PSTN through Opensips with mediaproxy
>> enabled. It works properly.
>>
>>
>> Calls with ring group enabled:
>> Incoming call from PSTN to asterisk through opensips with mediaproxy
>> enabled. The incoming call activate the asterisk's  ring group and
>> sends the call to sip phone and external line – outgoing pstn call.
>> Both devices ring simultaneous. When hang-up:
>>
>> A) soft phone, the signaling and media work properly.
>> B) External line, the signaling works properly but the media is not
>> open. The system (opensips/mediaproxy) generates two media
>> sessions(incoming and outgoing) but the ip of asterisk at both
>> sessions has value Unknown. The mediaproxy/opensips tries to connect
>> the two legs through asterisk. But this does not work because the
>> asterisk acts as packet2packet bridge.
>>
>>
>> Please, can you provide any help/sugestion about this problem?
>

> Ruud Klaver
> AG Projects