[OpenSIPS-Users] B2BUA marketting scenario

Bogdan-Andrei Iancu bogdan at opensips.org
Tue Sep 1 10:23:07 CEST 2015


Hi Sebastian,

Not yet, but I'm preparing the setup to run the test and fix. Anyhow, I 
haven't forgot about this !

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 31.08.2015 23:38, Sebastian Sastre wrote:
> Bodgan,
>
> Did you have a change to look into this? just curious to know if you 
> replicated the problem.
>
> thanks !
>
>
> On Tue, Aug 18, 2015 at 12:16 PM, Sebastian Sastre 
> <sastre.sebastian at gmail.com <mailto:sastre.sebastian at gmail.com>> wrote:
>
>     Bogdan,
>
>     it appears to be broken as of 1.11 and 2.1 yes. I couldn't find
>     any more indications in the logs that would point to a visible
>     error, but the ACK still has no SDP.
>
>     I have a few machines to test this out with the different
>     versions, let me know if you want a specific trace or core dump,
>     happy to help.
>
>     thanks !
>
>
>     On Tue, Aug 18, 2015 at 9:22 AM, Bogdan-Andrei Iancu
>     <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>
>         Sebastian,
>
>         So 1.11 and above are broken in this late ACK generation ? If
>         so, I will dig into .
>
>         Regards,
>
>         Bogdan-Andrei Iancu
>         OpenSIPS Founder and Developer
>         http://www.opensips-solutions.com
>
>         On 18.08.2015 16:20, Sebastian Sastre wrote:
>>         Bodgan,
>>
>>         Yes , i tried 1.11 and had the same issue, so i went down to
>>         1.8 TLS and it worked right away with the same scenario. A
>>         fee config changes but overal its the standrad script.
>>
>>         With 1.8 i see the sdp on the Ack and the call connects
>>         without problems. Even video.
>>
>>         Not sure why it did not work on higher versions.
>>
>>         Regards,
>>
>>
>>         On Tue, Aug 18, 2015 at 7:42 AM, Bogdan-Andrei Iancu
>>         <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>
>>             Hi Sebastian,
>>
>>             You mentioned yesterday on IRC channel that you fixed the
>>             problem ?
>>
>>             Regards,
>>
>>             Bogdan-Andrei Iancu
>>             OpenSIPS Founder and Developer
>>             http://www.opensips-solutions.com
>>
>>             On 17.08.2015 13:40, Sebastian Sastre wrote:
>>>             Bodgan,
>>>
>>>             Thanks i wasn't sure on the ack process. This is the log
>>>             , the scenario is triggered by a httpd json call.
>>>
>>>             INFO:b2b_logic:b2bl_add_client: adding entity
>>>             [0x7f718dfa7068]->[B2B.173.7331923] to tuple
>>>             [0x7f718dfa0cd0]->[685.0]
>>>             WARNING:b2b_logic:b2bl_delete_entity: entity
>>>             [0x7f718dfa2d18]->[] not found for tuple [685.0]
>>>             INFO:b2b_logic:b2bl_delete_entity: delete tuple [685.0],
>>>             entity []
>>>             INFO:b2b_logic:b2bl_add_client: adding entity
>>>             [0x7f718dfa4d28]->[B2B.173.5533781] to tuple
>>>             [0x7f718dfa0cd0]->[685.0]
>>>             INFO:b2b_logic:b2b_add_dlginfo: Dialog pair:
>>>             [B2B.173.7331923] - [B2B.173.5533781]
>>>
>>>             and the trace looks like this
>>>
>>>             172.10.1.21 -> 172.10.1.107 SIP 436 Request: INVITE
>>>             sip:sebas3 at 172.10.1.107:5060
>>>             <http://sip:sebas3@172.10.1.107:5060>
>>>             172.10.1.107 -> 172.10.1.21 SIP 346 Status: 100 Giving a try
>>>             172.10.1.107 -> 172.10.1.21 SIP 456 Status: 180 Ringing
>>>             172.10.1.107 -> 172.10.1.21 SIP/SDP 1088 Status: 200 Ok,
>>>             with session description
>>>
>>>             172.10.1.21 -> 172.10.1.20 SIP/SDP 843 Request: INVITE
>>>             sip:1 at 172.10.1.20:5060 <http://sip:1@172.10.1.20:5060>,
>>>             with session description
>>>             172.10.1.20 -> 172.10.1.21 SIP 390 Status: 100 Trying
>>>             172.10.1.20 -> 172.10.1.21 SIP/SDP 1252 Status: 200 OK,
>>>             with session description
>>>
>>>             172.10.1.21 -> 172.10.1.107 SIP 526 Request: ACK
>>>             sip:sebas3 at 73.139.116.217
>>>             <mailto:sip%3Asebas3 at 73.139.116.217>
>>>             172.10.1.21 -> 172.10.1.20 SIP 446 Request: ACK
>>>             sip:1 at 172.10.1.20:5060;transport=udp
>>>             <mailto:sip:1 at 172.10.1.20:5060;transport=udp>
>>>
>>>             172.10.1.107 -> 172.10.1.21 SIP 474 Request: BYE
>>>             sip:DialerProxy at 172.10.1.21:5060
>>>             <http://sip:DialerProxy@172.10.1.21:5060>
>>>             172.10.1.21 -> 172.10.1.20 SIP 446 Request: BYE
>>>             sip:1 at 172.10.1.20:5060;transport=udp
>>>             <mailto:sip:1 at 172.10.1.20:5060;transport=udp>
>>>             172.10.1.20 -> 172.10.1.21 SIP 545 Status: 200 OK
>>>             172.10.1.21 -> 172.10.1.107 SIP 629 Status: 200 OK
>>>
>>>             thanks !
>>>
>>>
>>>             On Mon, Aug 17, 2015 at 5:47 AM, Bogdan-Andrei Iancu
>>>             <bogdan at opensips.org <mailto:bogdan at opensips.org>> wrote:
>>>
>>>                 Hi Sebastian,
>>>
>>>                 The 200OK from FS must be followed by ACK+SDP to
>>>                 linphone. See:
>>>                 http://www.opensips.org/Documentation/Tutorials-B2BUA#toc14
>>>
>>>                 If this does not happen -> do you see any errors in
>>>                 the logs (around the processing of 200OK from FS) ?
>>>
>>>                 Regards,
>>>
>>>                 Bogdan-Andrei Iancu
>>>                 OpenSIPS Founder and Developer
>>>                 http://www.opensips-solutions.com
>>>
>>>                 On 17.08.2015 04:18, Sebastian Sastre wrote:
>>>>                 Hi guys,
>>>>
>>>>                 Im using the B2BUA module to send a call out to our
>>>>                 subscribers and bridge them with our IVR server on
>>>>                 answer.
>>>>
>>>>                 The subscriber side uses linphone and the media
>>>>                 server is a freeswitch 1.6. When placing the call
>>>>                 thru the trigger scenario MI command, the initial
>>>>                 invite does not have any SDP inside which makes sense.
>>>>
>>>>                 Once the 200ok is received from the linphone
>>>>                 client, opensips uses  the SDP contained in the 200
>>>>                 to generate an invite to the freeswitch box. which
>>>>                 is great.
>>>>
>>>>                 However, when the 200ok is received from
>>>>                 freeswitch, the following ACK back the linphone
>>>>                 client does not contain the SDP and Linphone
>>>>                 complains with "No codec intersection" and sends an
>>>>                 immediate bye.
>>>>
>>>>                 Am i right to think that the sdp should go in the
>>>>                 ack to create a late offer?
>>>>                 Should i be sending a re invite?
>>>>
>>>>                 any help appreciated.
>>>>
>>>>                 My scenario is simple.
>>>>
>>>>                 <?xml version="1.0"?>
>>>>                 <scenario id="dialer" name="MS start conditional"
>>>>                 param="2" type="extern">
>>>>                 <init>
>>>>                 <bridge>
>>>>                 <client>
>>>>                 <id>client1</id>
>>>>                 <destination>
>>>>                  <value type="param">1</value>
>>>>                 </destination>
>>>>                 </client>
>>>>                 <client>
>>>>                 <id>client2</id>
>>>>                 <destination>
>>>>                  <value type="param">2</value>
>>>>                 </destination>
>>>>                 </client>
>>>>                 </bridge>
>>>>                 <state>1</state>
>>>>                 </init>
>>>>                 </scenario>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>                 _______________________________________________
>>>>                 Users mailing list
>>>>                 Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>>>                 http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>
>
>

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