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

[Joshua Colp]

Johann Steinwendtner wrote:
> 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

The UnicastRTP dial string allows specifying the format. I did not touch 
MulticastRTP.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
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