[SR-Users] rtpproxy and kamailio doesnt work out from the box.

[Chandrakant Solanki]

Try to run rtpproxy on private ip not on local 127.0.0.1

On Tue, Jun 28, 2011 at 11:23 AM, MingHon <gminghon at gmail.com> wrote:

> Hi,
>
> i fixed the audio issue for 102 to 103 vice versa.
>
> by fixing the canreinvite in asterisk.
>
> from uac the rtp packet will route to kamailio den forward to asterisk.
>
> can we bypass the rtp packet going to asterisk?
>
> and here is the update for uac 101 issue.
>
> when 101 call to voicemail or 102/103 there is no audio.
>
> in wireshark i saw 101 send rtp packet to a private ip belong to asterisk.
>
> but if 102/103 call to 101 both uac got audio.
>
> i realize this is because 101 is the first uac registered before 102/103
> and because it did not have the received: field in ul show.
>
> please adv.
>
> --
> Regards,
>
> MingHon
>
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>


-- 
Regards,

Chandrakant Solanki
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