[SR-Users] bypass rtp traffic.

Carsten Bock carsten at ng-voice.com
Thu Jul 14 17:22:18 CEST 2011


Hi,

have you tried "canreinvite=yes" on your Asterisk-box?
If that does not help, there is probably no way to make the
RTP-Traffic bypass your asterisk box...

Carsten


2011/7/14 MingHon <gminghon at gmail.com>:
> Hello,
> anyone?
> currently my setup look like this.
> when UA1 call UA2 and the rtp traffic flow to kamailio and asterisk.
> [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]
>                                       ^
>                                       |
>                               RTP TRAFFIC
>                                       |
>                                       v
>                               [ASTERISK]
> what i need to achieve is UA1 call UA2 and rtp traffic flow to kamailio but
> not to asterisk.
> can kamailio handle the rtp traffic it own?
> [UA1] <--(rtp)--> [Kamailio/RTPProxy] <--(rtp)--> [UA2]
>                                      ^
>                                      |
>                                      X
>                                      |
>                                      v
>                              [ASTERISK]
>
> Thanks in advance.
> --
> Regards,
>
> MingHon
>



-- 
Carsten Bock
http://www.ng-voice.com
mailto:carsten at ng-voice.com

Schomburgstr. 80
22767 Hamburg
Germany

Mobile +49 179 2021244
Office +49 40 34927219
Fax +49 40 34927220

~~~~~
Checkout SIP-Provider CE:
http://www.sipwise.com/products/spce/overview/



More information about the sr-users mailing list