Hey Guys, I was on vacation when this thread happened but I'm also working on this now. The implementation uses the existing ARI channel and bridge recording endpoints ands add the ability to specify a URI in the form of (udp|tcp|tls)://hostname:port to stream the media. This makes use of the existing chan_bridge_media driver and only requires a scheme handler similar to Seán's res_audiosocket. This implementation is more targeted to real-time speech recognition/transcription/captioning and is therefore one way (outbound). A future enhancement is planned that would send each channel in a bridge as a separate audio channel in a multi-channel container.
I'm not suggesting that this should replace Seán's audiosocket stuff but I did want to let you know what was in the pipeline. george On Fri, Jul 5, 2019 at 7:38 AM Seán C. McCord <ule...@gmail.com> wrote: > Solutions such as Jack are non-network oriented and severely limited in > scalability. There are, of course, many other options, but the closest are > chan_rtp and chan_nbs. RTP is a good option except for the difficulty for > non-telephony people to interact with it. NBS is deprecated, undocumented, > and unsupported by any locatable resources. > > While the original app interface from last year required dialplan, the > channel interface does not. It is a plain channel which can be used by ARI > directly. > > > On Fri, Jul 5, 2019, 15:28 Sylvain Boily <sylv...@wazo.io> wrote: > >> Hello Seán, >> >> On 2019-07-05 4:45 a.m., Seán C. McCord wrote: >> >> A brief update: >> >> I have adapted my app_audiosocket from last year to become >> chan_audiosocket, a full bidirectional audio channel interface for Asterisk >> to any AudioSocket service (which itself is a dead-simple implementation). >> I'll be demoing it in my talk next week at CommCon, for anyone who might be >> interested. I'm going to try to have it ready to push to gerrit for review >> this weekend. >> >> >> I'll be there. >> >> >> For now, you can see it in the 'channel' branch of >> github.com/CyCoreSystems/audiosocket. >> >> >> This is very different from what we did. You need dialplan to use it. In >> our case we don't need any dialplan to use it, it's more ARI approach. >> >> Sylvain >> > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev -- *George Joseph* Digium - A Sangoma Company | Software Developer | Software Engineering 445 Jan Davis Drive NW - Huntsville, AL 35806 - US direct/fax: +1 256 428 6012 Check us out at: https://digium.com · https://sangoma.com
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