[Freeswitch-dev] Freeswitch chooses the first codec and ignores the rest

Wed Sep 25 16:57:25 MSD 2013

Hi,
I will start to work on video transcoding as well

transcoding code could be added to video_bridge_thread ( in the core,
keeping consistency with audio_bridge_thread)
Or
It could trigger an application module that would be responsible for video
transcoding (modularity)

On Mon, Sep 23, 2013 at 4:05 PM, Seven Du <dujinfang at gmail.com> wrote:

> Sam your code is a good start and I'm sure you'll get how FreeSWITCH is
> built soon. Let's find a time talk.
>
> --
> Seven Du
> http://www.freeswitch.org.cn
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> http://www.dujinfang.com
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Monday, September 23, 2013 at 4:05 PM, Sam Russell wrote:
>
> I'd be keen for that. I'm not a software engineer, but I've managed to get
> libavcodec transcoding VP8-H264 from file to file (
> https://github.com/samrussell/vodec) - I'd just need to get my head
> around how freeswitch is built.
>
>
> On Sat, Sep 21, 2013 at 6:58 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> Yes Seven Du and I are discussing this project right now.
> Video codecs will require some more core functionality to perform the
> actual transcoding etc but yes we are interested in this.
> Maybe we can setup a discussion.
>
>
>
> On Thu, Sep 19, 2013 at 9:24 PM, Sam Russell <sam.h.russell at gmail.com>wrote:
>
> Thanks for this Anthony - I'm still on the stable branch which doesn't
> seem to support this, but I'll get it upgraded to HEAD and test this & your
> new patch too.
>
> On the topic of codecs, has there been much thought given to video
> transcoding? I've seen references to a "mod_ffmpeg" floating around, and
> I've done a bit of transcoding work with libavcodec myself - it seems
> doable, and the little bit that I've looked at the freeswitch codec
> examples looks really extensible. If I were to write codecs for H264 and
> VP8 (and maybe H263+ too), would they be something that you'd try to
> incorporate into the main project, or would it be better to have as a
> third-party mod due to the extra hassle of having to build with
> libav(codec|format|util)?
>
>
> On Fri, Sep 20, 2013 at 9:18 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> I'd say its relatively hard.  Easier in master which is the only place it
> would be possible to do but it requires some heavy mods to the media engine
> and the rtp stack to track all the payloads and switch between them.
>
> I do think there is another workaround if you set this profile param:
>
> rtp-autofix-pt=true
>
>
> That will detect when its not the one FS thinks and it will reinit the
> codec.
>
>
>
>
>
>
>
> On Thu, Sep 19, 2013 at 4:10 PM, Sam Russell <sam.h.russell at gmail.com>wrote:
>
> Thanks for the quick reply Anthony.
>
> Do you think this is a hard/easy thing to implement? I'm keen to
> contribute a bit to freeswitch, and this is functionality that I'd like to
> write some code for, and I can do what I like with our office PBX so I can
> test it here too.
>
>
> On Fri, Sep 20, 2013 at 8:56 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> We currently only support one codec in 200OK  The workaround for this is
> to only offer it PCMU so it has no choice.
>
>
>
> On Thu, Sep 19, 2013 at 2:52 PM, Sam Russell <sam.h.russell at gmail.com>wrote:
>
> I've had a couple of calls recently that have been a little weird -
> here's the scenario:
>
> A (local freeswitch) sends invite with PCMU, PCMA as codecs
>
> B (foreign) sends 200 OK with PCMA, PCMU as codecs
>
> Freeswitch sees the response and sets audio codec to PCMA, but other
> end sends us PCMU (as it was our first choice in this case - sometimes
> they choose a random one that both of us support).
>
> RTP is fine with this - each packet refers to the # of the codec
> (either standard or a reference to an rtpmap from the SDP), but
> freeswitch doesn't like it - it forwards the PCMA from my soft phone,
> but the PCMU coming in from the foreign phone gets dropped by
> freeswitch and doesn't get forwarded onto my soft phone (packet trace
> shows freeswitch being the black hole).
>
> Debug output shows freeswitch setting the audio codec, and I suspect
> this means it then only accepts RTP with that codec, instead of
> handling all codecs that it offered in its invite.
>
> Is this expected behaviour? I went trawling through RFCs yesterday and
> nothing jumped out at me, and I'm seeing this on a couple of call
> managers - also sometimes the other end will send H263+ even though
> both ends offered H264 as preferred, and freeswitch drops this in the
> same manner. I think freeswitch should be handling all codecs that it
> offers, especially for video when it only does pass through, so
> spinning up instances of extra "codecs" shouldn't create much
> overhead.
>
> Sent from my iPhone
>
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>
> --
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>
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> --
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>
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> _________________________________________________________________________
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>
> 
> 
>
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> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
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> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
> _________________________________________________________________________
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> http://www.freeswitchsolutions.com
>
> 
> 
>
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>
> 
> 
>
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> 
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