[asterisk-dev] [Code Review] 3981: chan_rtp: Add unicast RTP support to chan_multicast_rtp.

[Johann Steinwendtner]

On 2014-09-07 17:07, Joshua Colp wrote:
> This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3981/
>
>

>   Testing
>
> Originated a call to a UnicastRTP channel and sent it to a Playback. Confirmed that RTP was sent to the provided IP address/port with the given format.
>

Hello, can you please explain what you mean by "with the given format".
There is a patch from John R. Covert which adds the capability of selecting the codec. Or is this not necessary in trunk.

http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.devel/48495

Thanks

Regards,

Hans