<div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Apr 6, 2016 at 11:47 AM, Vincent Gire <span dir="ltr"><<a href="mailto:vincent.gire@gmail.com" target="_blank">vincent.gire@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr"><br><div class="gmail_extra"><br><div class="gmail_quote"><span class="">On Wed, Apr 6, 2016 at 6:10 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Firstly,<div><br></div><div>You cannot stream WAV. Its not possible. WAV is not a streaming format, its a container. Its basically a raw audio data stream with a header explaining the characteristics of the stream.</div></div></blockquote><div><br></div></span><div>Yes. Sorry for the language abuse and thank you for clearing this out.</div><span class=""><div> </div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div><br></div><div><br></div><div>Next,</div><div><br></div><div>You need to compare your specific needs to the goal of the software to be flexible and usable to all.</div><div><br></div><div>The reason the speech interface was suggested is that the point of the speech interface is precisely to tap into live audio streams and perform asynchronous ASR operations while the original call does whatever it wants like talk to another caller or listen to a file playing etc.</div><div><br></div><div>What you envision is more of a synchronous single instruction to record a file in realtime to the remote server over httapi.</div></div></blockquote><div><br></div></span><div>Exactly.</div><span class=""><div> <br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div>The considerations to implement this would be what format to send the audio in. The source audio could be any combination of sample rate, channel count and encoding format.</div><div>To record we would translate the encoded format into signed linear at the original input sample rate and channels. These variances must be communicated before you start sending the data so it would be a matter of figuring out an existing standard or implement a proprietary one.</div></div></blockquote><div><br></div></span><div>I am not sure I am following here.</div><div>I might again have misused the word streaming.</div><div>I was thinking that when we specify a http target for a record in mod_httapi, it could start sending the corresponding file (WAV or other) to the HTTP server as chunked data as soon as the record starts.</div><div>In the case of a WAV target, the WAV header would be sent first as part of the first HTTP data chunks and the raw audio data stream would follow in the next data chunks. </div><div><div class="h5"><div>Does it make sense ?</div></div></div></div></div></div></blockquote><div><br></div><div><br></div><div>I agree that you are not following as I gave a relatively lengthy and specific response. I know exactly what you are asking for so there is no need to re-explain.</div><div><div>The way you ask for it is very specific to your own needs so it would not make a very useful feature in FS.</div></div><div><br></div><div>Are you asking for someone to make this feature for you or are you asking for advice on how you can implement it?</div><div>I think you are underestimating the complexity and potential scalability of the overall project you are doing (not just the FS portion) and you should review your entire architecture.<br></div><div><br></div><div><br></div><div> <br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-style:solid;border-left-color:rgb(204,204,204);padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div class="gmail_quote"><div><div class="h5"><div><br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div><br></div><div><br></div><div><br></div><div><br></div></div><div class="gmail_extra"><div><div><br><div class="gmail_quote">On Wed, Apr 6, 2016 at 10:42 AM, Vincent Gire <span dir="ltr"><<a href="mailto:vincent.gire@gmail.com" target="_blank">vincent.gire@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">We already handle the multiple providers on the HTTP server.<div>So you would recommend to implement a module against the speech interface streaming to the HTTP server ?</div><div>Wouldn't it be easier (and less redundant) to add streaming capabilities to mod_httpapi ?</div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Apr 6, 2016 at 12:22 AM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div style="word-wrap:break-word">The best way to handle this is probably to write a custom module against the speech interface that sends the streams like you are describing, and supports multiple providers. How exactly do you go about sending to multiple at the same time and combining the results, as this might actually be the trickiest part as youll need to integrate that into a module that uses the speech interface to be able to at all sanely handle that in freeswitch<div><div><div><br><div><blockquote type="cite"><div>On Apr 5, 2016, at 1:24 PM, Vincent Gire <<a href="mailto:vincent.gire@gmail.com" target="_blank">vincent.gire@gmail.com</a>> wrote:</div><br><div><div dir="ltr">We use multiple ones to compare confidence.<div>They all support chunk transfer and provide a HTTP transaction similar to :</div><div><a href="http://developer.att.com/apis/speech/docs/v3" target="_blank">http://developer.att.com/apis/speech/docs/v3</a><br></div><div>Look at 3/ Make API Calls</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 5, 2016 at 7:14 PM, Michael Jerris <span dir="ltr"><<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">Unimrcp isn't going to provide the interface he's talking about. Can I ask which engine this is?<div><div><br><br>On Tuesday, April 5, 2016, Sergey Safarov <<a href="mailto:s.safarov@gmail.com" target="_blank">s.safarov@gmail.com</a>> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><p dir="ltr">Look at <a href="http://www.unimrcp.org/" target="_blank">http://www.unimrcp.org/</a> and <a href="https://wiki.freeswitch.org/wiki/Mod_unimrcp" target="_blank">https://wiki.freeswitch.org/wiki/Mod_unimrcp</a></p>
<br><div class="gmail_quote"><div dir="ltr">On Tue, Apr 5, 2016, 19:38 Vincent Gire <<a>vincent.gire@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div>We are building an IVR completely driven by ASR.</div><div>ASR is performed in a distant location by a HTTP service (supporting chunked transfer) and adds an incompressible latency. We would like to stream the record to the ASR service as soon as it starts to reduce the overall latency before response.</div><div>Does it make sense ?</div><div><br></div></div><div class="gmail_extra"></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 5, 2016 at 5:39 PM, Michael Jerris <span dir="ltr"><<a>mike@jerris.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div style="word-wrap:break-word">I would stay away from mod_vlc. Its audio portions with recording have known issues. We do use the rtmp streaming in mod_av heavily but thats obviously not wav. Can you explain a bit more why you have this requirement?<div><div><div><br><div><blockquote type="cite"><div>On Apr 5, 2016, at 5:36 AM, Vincent Gire <<a>vincent.gire@gmail.com</a>> wrote:</div><br><div><div dir="ltr">Ok thanks.<br>It looks promising !<div>I'll dig into mod_vlc.</div><div><br></div><div>Best </div><div><br></div><div>Vincent</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Apr 5, 2016 at 10:04 AM, Sergey Safarov <span dir="ltr"><<a>s.safarov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Think is requred streaming feature of freeswitch.<div>Look at <a href="https://freeswitch.org/confluence/display/FREESWITCH/mod_esf" target="_blank">mod_esf</a> and <a href="https://wiki.freeswitch.org/wiki/Mod_vlc" target="_blank">mod_vlc</a></div><div>Instruction to compile mod_vlc on provided link is to old but helpfull to undestand how to stream media to http server.</div><div><br></div><div>For compiling mod_vlc please use <a href="https://freeswitch.org/stash/projects/SD/repos/vlc/browse" target="_blank">vlc repo</a> and <a href="https://freeswitch.org/confluence/display/FREESWITCH/Creating+RPM+repositary" target="_blank">centos instruction</a>.</div><div>After you intall vlc, then you can enable mod_vlc module in freeswitch sources(SPEC file) and compile freeswitch.</div><div><br></div><div>Sergey</div><div><br></div></div><br><div class="gmail_quote"><div dir="ltr">вт, 5 апр. 2016 г. в 10:31, Vincent Gire <<a>vincent.gire@gmail.com</a>>:<br></div><div><div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">webdav, mod_http_cache or mod_httapi all results in sending the recording only <b>after</b> it is complete.<div>They all write the <span style="font-family:sans-serif;font-size:12.8px;line-height:19.2px">recording to a file, wait for the recording to complete and the file to close and then send it over HTTP.</span></div><div><font face="sans-serif"><span style="font-size:12.8px;line-height:19.2px"><br></span></font><div>I would like to start sending the recording to the remove server as soon as it starts (max 1 sec latency).</div><div>mod_http_cache or mod_httapi would be perfect if they were streaming the recording like mod_shout.</div><div><br></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On Mon, Apr 4, 2016 at 8:50 PM, Sergey Safarov <span dir="ltr"><<a>s.safarov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><p dir="ltr">Input/output latency is not problem. I use Kazoo on my servers and call recording is stored to database during 5 seconds after hangup.<br>
What is broken in your case if save file using webdav or http_cache?</p><div><div>
<br><div class="gmail_quote"><div dir="ltr">On Mon, Apr 4, 2016, 21:10 Vincent Gire <<a>vincent.gire@gmail.com</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">Hello Sergey,<div><br></div><div>Thank you for your answer.</div><div>I've looked into webdav mounted filesystem.</div><div><br></div><div>Unfortunately, most WebDav clients (especially davfs2 on debian) do a lot of buffering, caching and even lock-null requests (lock a non existent resource before writing to it). I also suspect that they wait for the end of the write operation.</div><div>The result is a latency of a few seconds witch is not much better than what I achieve with mod_shout if I transcode the MP3.</div><div><br></div><div>Any other idea ?</div><div><br></div><div>Thank you !</div><div><br></div><div>Best regards</div><div><br></div><div>Vincent</div><div><div class="gmail_extra"></div></div></div><div dir="ltr"><div><div class="gmail_extra"><br><div class="gmail_quote">On Sun, Apr 3, 2016 at 7:30 PM, Sergey Safarov <span dir="ltr"><<a>s.safarov@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><p dir="ltr">Please look at webdav mounted filesystem.</p>
<br><div class="gmail_quote"><div><div><div dir="ltr">On Sun, Apr 3, 2016, 19:17 Vincent Gire <<a>vincent.gire@gmail.com</a>> wrote:<br></div></div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div dir="ltr"><div><div><span style="white-space:pre-wrap">Hi all,</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">Thank you to all contributing to FreeSWITCH !</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">I'm working on a IVR project where logic is implemented on a HTTP server.</span></div><div><span style="white-space:pre-wrap">We are leaving Twilio because we now need to record and live stream the session to the HTTP server in WAV format (chunked transfer encoding).</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap"><b>mod_httapi</b> looks great (HT TAPI very similar to Twilio's) but it seems that the records are first saved to disk before there are sent to the server as chunked data. </span></div><div><span style="white-space:pre-wrap">We need the transfer to start as soon as the recording starts.</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap"><b>mod_shout</b> does start the request almost as the records starts but it does not support WAV file and shout:// is not exactly a HTTP request (SOURCE method instead of PUT).</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">Is there a way to use these modules to achieve our goal ?</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">If not, we are willing to author a specific module or rather contribute to the existing ones.</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">We've identified two approaches:</span></div><div><ol><li><span style="white-space:pre-wrap">From <b>mod_httapi</b>
</span>Modify mod_httapi to directly stream the record instead of completely saving it to disk before the HTTP chunked transfer starts.<br>This seems the most logical but with more than 3000 lines, mod_httapi does not seem to be the easiest module to build upon for newcomers!<br><span style="white-space:pre-wrap"><br></span></li><li>From <b>mod_shout</b></li><ol><li>Modify libshoot to replace the custom SOURCE method with standard HTTP PUT method</li><li>Modify mod_shout to support wav files</li><li>Implement our IVR in script (javascript/lua)</li></ol></ol></div><div><span style="white-space:pre-wrap">What do you think ?</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">Thank you for your help.</span></div><div><span style="white-space:pre-wrap"><br></span></div><div><span style="white-space:pre-wrap">Cheers,</span></div></div></div></div></div></blockquote></div></blockquote></div></div></div></div></blockquote></div></div></div></blockquote></div></div></blockquote></div></div></div></blockquote></div></div></div></blockquote></div></div></div></div></div><br>_________________________________________________________________________<br>
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UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br></div></div><div><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ <a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a> </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div></div>
</div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org" target="_blank">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org" target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div></div></div><span class=""><font color="#888888"><br><br clear="all"><div><br></div>-- <br><div><div dir="ltr">Vincent Gire<br></div></div>
</font></span></div></div>
<br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" rel="noreferrer" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://confluence.freeswitch.org" rel="noreferrer" target="_blank">http://confluence.freeswitch.org</a><br>
<a href="http://www.cluecon.com" rel="noreferrer" target="_blank">http://www.cluecon.com</a><br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" rel="noreferrer" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" rel="noreferrer" target="_blank">http://www.freeswitch.org</a><br></blockquote></div><br><br clear="all"><div><br></div>-- <br><div class="gmail_signature"><div dir="ltr"><div><div dir="ltr"><div><div dir="ltr">Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬<div><br><div>☞ <a href="http://freeswitch.org/" target="_blank">http://freeswitch.org/</a> ☞ <a href="http://cluecon.com/" target="_blank">http://cluecon.com/</a> ☞ <a href="http://twitter.com/FreeSWITCH" target="_blank">http://twitter.com/FreeSWITCH</a></div><div><div>☞ <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch ☞ <u><a href="http://freeswitch.org/g+" target="_blank">http://freeswitch.org/g+</a></u><br><br></div><div>ClueCon Weekly Development Call <br></div><div>☎ <a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a> ☎ +19193869900 </div><div><br></div></div></div><div><a href="https://www.youtube.com/watch?v=9XXgW34t40s" style="color:rgb(17,85,204);font-size:12.8px" target="_blank">https://www.youtube.com/watch?v=9XXgW34t40s</a></div><div><a href="https://www.youtube.com/watch?v=NLaDpGQuZDA" target="_blank">https://www.youtube.com/watch?v=NLaDpGQuZDA</a><br></div></div></div></div></div></div></div>
</div></div>