[OpenSIPS-Users] B2BUA marketting scenario

Bogdan-Andrei Iancu bogdan at opensips.org
Tue Aug 18 15:22:27 CEST 2015


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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