asterisk users - Jul 2015 - How to create direct media with PJSIP.conf configurations in Asterisk 13?

Dear Asterisk-Users,


By means of Asterisk 11 and sip.conf, I got success implementing early media.
That is, all information that come from callee (SIP 183 message/ SDP) is passed
to the caller without any modification in the SDP body.


However, in Asterisk 13 and using pjsip.conf I'm still failing to do the
same thing. See:




Softphojne1 <------------------------------------------------> Asterisk
<--------------------------------------->Softphone2


        |  ----------------------SIP INVITE ------------------------>|

                                                                                
| ----------------------------SIP INVITE------------>|



                                                .                               
.

                                                .                               
.

                                                .                               
.

                                                .                               
.

                                                                                
|<----------SIP 183 ---------------------------------|

                                                                                
SDP : Media Description, name and address (m): audio 4000 RTP/AVP 8 96

                                                                                
Media Description, name and address (m): video 5000 RTP/AVP 97

        |<--------------------------SIP 183-----------------------------|

           SDP :Media Description, name and address (m): audio 13258 RTP/AVP 8
96

                      Media Description, name and address (m): video 16002
RTP/AVP 97



So, there is bit modification in SDP body, caused by Asterisk. As long as
I'm intending to implement direct media, I believe that Asterisk 13 has some
special configuration to be done in PJSIP.conf file, that will allow things work
very well again, as in Asterisk 11 and sip.conf.


How to configure pjsip.con file or Asterisk, to run direct-media? Or , where to
find a tutorial about it on Internet?


Any hint will be very helpful!


Thanks a lot.






RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAl 979   (Brasil)
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Rodrigo Pimenta Carvalho wrote:
> Dear Asterisk-Users,
>
>
> By means of Asterisk 11 and sip.conf, I got success implementing early
> media. That is, all information that come from callee (SIP 183 message/
> SDP) is passed to the caller without any modification in the SDP body.
PJSIP does not support early direct media. It is only once the call has 
been established that it will be done.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

Thank you Joshua!
In this case I finally decide to use SIP Proxy. I have to start testing the SIP
Proxy today.

Best Regards.

RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979   (Brasil)
________________________________________
De: asterisk-users-bounces at lists.digium.com <asterisk-users-bounces at
lists.digium.com> em nome de Joshua Colp <jcolp at digium.com>
Enviado: quinta-feira, 16 de julho de 2015 10:54:56
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [asterisk-users] How to create direct media with PJSIP.conf
configurations in Asterisk 13?

Rodrigo Pimenta Carvalho wrote:
> Dear Asterisk-Users,
>
>
> By means of Asterisk 11 and sip.conf, I got success implementing early
> media. That is, all information that come from callee (SIP 183 message/
> SDP) is passed to the caller without any modification in the SDP body.
PJSIP does not support early direct media. It is only once the call has
been established that it will be done.

--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

--
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