It may not be suitable for your use case, but you could instantiate a
UnicastRTP channel. It will allocate an RTP port and store it into a
channel variable.
Jean
Le 22/07/2019 à 10:01, Dan Jenkins a écrit :
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
<mailto: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
<mailto: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 <mailto: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 <mailto: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 <mailto: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
<http://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
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CyCore Systems
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