[Asterisk-code-review] multicast RTP: Add dialing options (asterisk[master])

[Mark Michelson]

Hello Richard Mudgett, Anonymous Coward #1000019,

I'd like you to reexamine a change.  Please visit

    https://gerrit.asterisk.org/2910

to look at the new patch set (#3).

Change subject: multicast RTP: Add dialing options
......................................................................

multicast RTP: Add dialing options

This adds a new parameter to the end of a multicast RTP dialing string.
This parameter defines the following options:

* i: Set the interface from which multicast RTP is sent
* l: Set whether multicast packets are looped back to the sender
* t: Set the TTL for multicast packets
* c: Set the codec to use for RTP

ASTERISK-26068 #close
Reported by Mark Michelson

Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
---
M channels/chan_rtp.c
A include/asterisk/multicast_rtp.h
M res/res_rtp_multicast.c
A res/res_rtp_multicast.exports.in
4 files changed, 264 insertions(+), 13 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/10/2910/3
-- 
To view, visit https://gerrit.asterisk.org/2910
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219
Gerrit-PatchSet: 3
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>