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

[Joshua Colp]

Paul Belanger wrote:
>
> On Sep 7, 2014 2:28 PM, "Joshua Colp" <jcolp at digium.com
> <mailto: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.

UnicastRTP/<ip address>:<port>/<optional engine>/<optional format>

Good point though - I don't think we have a good mechanism for 
documenting dial strings.

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
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