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pimenta at inatel.br
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 Posted: Thu Jul 16, 2015 8:51 am Post subject: [asterisk-users] How to create direct media with PJSIP.conf |
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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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jcolp at digium.com
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| Posted: Thu Jul 16, 2015 8:55 am Post subject: [asterisk-users] How to create direct media with PJSIP.conf |
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Rodrigo Pimenta Carvalho wrote:
| Quote: | 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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pimenta at inatel.br
Guest
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| Posted: Thu Jul 16, 2015 9:22 am Post subject: [asterisk-users] How to create direct media with PJSIP.conf |
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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@lists.digium.com <asterisk-users-bounces@lists.digium.com> em nome de Joshua Colp <jcolp@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:
| Quote: | 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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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