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

[Paul Belanger]

On Sep 7, 2014 2:28 PM, "Joshua Colp" <jcolp at digium.com> wrote:
>
> 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.
>
What does the dial string look like? I didn't see any  documentation on it.
Mind you I am using my phone for the code review.
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