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Updated for 12.2.0 #7

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8d18dac
Updated patch for Asterisk 12.2.0
Thulinma May 10, 2014
1a49176
Updated readme for 12.2.0 version
Thulinma May 10, 2014
2a33f72
Updated readme some more.
Thulinma May 10, 2014
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16 changes: 7 additions & 9 deletions README.md
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Original file line number Diff line number Diff line change
@@ -1,14 +1,14 @@
#Asterisk Opus/VP8 patch
#Asterisk Opus patch
=======================

Since Opus and VP8 cannot, as of now, be integrated in the Asterisk repositories (learn why in this [thread](http://lists.digium.com/pipermail/asterisk-dev/2013-May/060356.html)), we prepared a patch that adds support for both codecs (Opus transcoding, VP8 passthrough) to [Asterisk 11.1.2](http://downloads.asterisk.org/pub/telephony/asterisk/releases/).
Since Opus cannot, as of now, be integrated in the Asterisk repositories (learn why in this [thread](http://lists.digium.com/pipermail/asterisk-dev/2013-May/060356.html)), we prepared a patch that adds support for Opus transcoding to [Asterisk 12.2.0](http://downloads.asterisk.org/pub/telephony/asterisk/releases/).

##Installing the patch
To support Opus, you'll need to install [libopus](http://www.opus-codec.org/downloads/) first. No library is needed for VP8, as its support is passthrough only.
To support Opus, you'll need to install [libopus](http://www.opus-codec.org/downloads/) first.

The patch was built on top of Asterisk 11.1.2: applying it on different versions may or may not work out of the box, but solving conflicts shouldn't be too hard anyway. Copy it in the Asterisk source folder and apply it:
The patch was built on top of Asterisk 12.2.0: applying it on different versions may or may not work out of the box, but solving conflicts shouldn't be too hard anyway. Copy it in the Asterisk source folder and apply it:

patch -p1 -u < asterisk_opus+vp8.diff
patch -p1 -u < asterisk_12.2.0_opus.diff

Run the bootstrap script to regenerate the configure:

Expand All @@ -18,7 +18,7 @@ Configure the patched Asterisk.

./configure --prefix=/usr

Make sure that codec\_opus and format\_vp8 are enabled in menuselect before going on. Besides, for better results, install the slin16 versions of the Asterisk sounds, which are not enabled by default.
Make sure that codec\_opus is enabled in menuselect before going on.

make menuselect

Expand All @@ -33,10 +33,8 @@ You can test Opus using the free softphone [PhonerLite](http://phonerlite.de/dow
Usage: opus set debug {status|none|normal|huge}
Enable/Disable Opus debugging: normal only debugs setup and errors, huge debugs every single packet

For VP8 you can make use of the open source softphone [Linphone](http://www.linphone.org/eng/linphone/news/linphone-3.5.0-released-for-desktop.html), which added support for VP8 in version 3.5.0.

##What is missing
SDP fmtp parameters related to Opus and defined in [draft-ietf-payload-rtp-opus](http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00) are parsed but currently ignored: this means that there's no interaction between chan\_sip and codec\_opus in that sense. Besides, there still is no ad-hoc Opus configuration file for codec defaults. VP8, as anticipated, is passthrough only: besides, there's currently no way to read VP8 files for Playback.
SDP fmtp parameters related to Opus and defined in [draft-ietf-payload-rtp-opus](http://tools.ietf.org/html/draft-ietf-payload-rtp-opus-00) are parsed but currently ignored: this means that there's no interaction between chan\_sip and codec\_opus in that sense. Besides, there still is no ad-hoc Opus configuration file for codec defaults.

##Help us improve the support!
Found an issue? Solved one? Added something that was missing? Help us make it better!
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