Also coming back to this with some real-life case issues I'm currently facing and why I can't use audiosocket :(
I need to be able to ask the ARI/AGI/AMI for an IP/port combo and for an external app to then connect into asterisk rather than asterisk connecting to a URI elsewhere. Lets say I already have a nodejs (or any other language) process taking care of controlling that call via ARI or even AGI (all the different ways) - I need that same process to handle the media I'm sending and receiving to/from asterisk and so if I already have that process up and then Asterisk calls out to a generic URI then that media will never make it back to the original nodejs process. I think its of upmost importance that I be able to ask asterisk for a host:port pair and then be able to connect to that port from my external application. What do people think? I thought we'd talked about this mechanism at devcon? Dan On Sat, Jul 20, 2019 at 2:39 PM Dan Jenkins <d...@nimblea.pe> wrote: > Just going to chime in and say I don't see a one way audio solution as > enough and I'd worry that doing that would lead to "oh but only so many > people need to chuck audio in". This has been discussed at at least 3 > devcons now. > > On Thu, Jul 18, 2019 at 2:09 PM Seán C. McCord <ule...@gmail.com> wrote: > >> I certainly don't mind if a better-designed system comes along and >> obviates my AudioSocket implementation. I built it because I needed it. >> However, bidirectional audio flow is critical for me (speech synthesis, >> external interfacing, real-time processed audio, processed injections, >> etc). While I would actually prefer a system which was a bit beefier than >> my own (simple protocol aside, there's a good deal of range between my >> protocol and MRCP), my meagre C skills (and patience) don't allow me to >> venture forth into those difficult waters. >> >> I do like the separate connection (unlike Wazo's) and the support of TLS >> (unlike mine)... and yours is certainly (even without looking) more >> performant. Mine also probably needs a multi-threaded, dedicated-receiver >> approach like most of the other channels which handle socket-received >> media, rather than the simple non-blocking I/O with null frame insertion. >> No perfect solution yet. >> >> >> >> On Thu, Jul 18, 2019 at 8:01 AM George Joseph <gjos...@digium.com> wrote: >> >>> 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 >>> >>> >> >> -- >> Seán C. McCord >> ule...@gmail.com >> CyCore Systems >> -- >> _____________________________________________________________________ >> -- 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 > >
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