I used MRCPv1 with remote servers. 3 MRCP servers handling requests from 12 FS servers. <br><br>Start with the example nuance MRCPv1 configuration in conf/mrcp_profiles and replace the rtp-ip, server-ip, and server-port. Use IP address, not hostnames, not "auto". If that doesn't work, reply with your configuration.<div>
<br></div><div>Chris<br><div><br><br><div class="gmail_quote">On Tue, Feb 26, 2013 at 2:33 PM, Jason Moran <span dir="ltr"><<a href="mailto:jmoran@secureachsystems.com" target="_blank">jmoran@secureachsystems.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple">
<div>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">After trying MRCPv1 – I don’t think anybody out
there is using Nuance for TTS remotely / externally! The audio is always
blank / silence(both Nuance and FreeSWITCH seem to think it is working ok). I
can get it to work with Nuance on localhost/<a href="http://127.0.0.1" target="_blank">127.0.0.1</a>, but I believe there is a
problem with FreeSWITCH/UniMRCP determining the correct rtp-ip when in MRCPv1
mode when using a server other than localhost. As shown in earlier posts in this
thread, I can get it to work with MRCPv2 on the same external server.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Regardless of all of these settings (client-ext-ip , client-ip ,
rtp-ip , rtp-ext-ip being set to an IP address other than 127.0.0.1, RTP is
still trying to use 127.0.0.1, although RTSP is using the external IP address
provided.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">mpf_rtp_stream.c from uniMRCP returns several references to
127.0.0.1 although it should not:<br>
mpf_rtp_stream.c:495 () Close RTP Receiver <a href="http://127.0.0.1:4018" target="_blank">127.0.0.1:4018</a> <- <a href="http://10.0.0.67:7914" target="_blank">10.0.0.67:7914</a>
[r:0 l:0 j:0 d:0 i:0]<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">and<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">mpf_rtp_stream.c:1294 () Send Compound RTCP Packet [BYE] [76
bytes] <a href="http://127.0.0.1:4019" target="_blank">127.0.0.1:4019</a> -> <a href="http://10.0.0.67:7915" target="_blank">10.0.0.67:7915</a><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">mpf_rtp_stream.c:1306 () Failed to Send Compound RTCP Packet
[BYE] [0 bytes] <a href="http://127.0.0.1:4019" target="_blank">127.0.0.1:4019</a> -> <a href="http://10.0.0.67:7915" target="_blank">10.0.0.67:7915</a><u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">I believe this explains why I cannot hear the audio that is
generated, as it should be using a 10.0.0.x IP address to receive and sent over
RTP. To be safe I also changed the external_rtp_ip in vars.xml, but that
hasn’t fixed the issue either.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Does anybody know where mpf_rtp_stream.c is getting this RTP IP
Address from? Why does it insist on 127.0.0.1 despite all settings saying
otherwise?<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">Jason<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p>
<div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> Christopher Rienzo
[mailto:<a href="mailto:cmrienzo@gmail.com" target="_blank">cmrienzo@gmail.com</a>] <br>
<b>Sent:</b> Monday, February 25, 2013 8:34 PM<br>
<b>To:</b> FreeSWITCH Users Help<br>
<b>Subject:</b> Re: [Freeswitch-users] Nuance Speech Server / Vocalizer
MRCPv2mod_unimrcp simultaneous licenses not releasing quickly enough.<u></u><u></u></span></p>
</div><div><div class="h5">
<p class="MsoNormal"><u></u> <u></u></p>
<p class="MsoNormal">You can also try MRCPv1 (RTSP) and see if there is better
performance. Using SIP to set up an MRCP session is silly.<u></u><u></u></p>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">Chris<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
<div>
<p class="MsoNormal">On Mon, Feb 25, 2013 at 6:42 PM, Michael Collins <<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>>
wrote:<u></u><u></u></p>
<p class="MsoNormal">Another case of DRM hurting only those who've paid for it...
*sigh*<br>
<br>
I recommend calling your VAR or Nuance and gently but firmly insisting on a resolution.
<br>
<span style="color:#888888"><br>
<span>-MC</span></span><u></u><u></u></p>
<div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><u></u> <u></u></p>
<div>
<p class="MsoNormal">On Mon, Feb 25, 2013 at 2:21 PM, <<a href="mailto:cmrienzo@gmail.com" target="_blank">cmrienzo@gmail.com</a>>
wrote:<u></u><u></u></p>
<div>
<div>
<p class="MsoNormal">We also had an issue with nuance eating licenses. We
had enough room to tolerate this but did have to occasionally reboot servers
when too many were consumed. <u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal">Chris<u></u><u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal"><u></u> <u></u></p>
</div>
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
On Feb 25, 2013, at 3:21 PM, Rob Hutton <<a href="mailto:justlikeef@gmail.com" target="_blank">justlikeef@gmail.com</a>>
wrote:<u></u><u></u></p>
</div>
<blockquote style="margin-top:5.0pt;margin-bottom:5.0pt">
<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">We have a similar situation on
a Cisco UCCX platform that we ended up having to write a script to bounce the
services every couple of hours to clear. We haven't seen a fix from
Nuance yet.<u></u><u></u></p>
<div>
<p class="MsoNormal">On Mon, Feb 25, 2013 at 2:32 PM, Jason Moran <<a href="mailto:jmoran@secureachsystems.com" target="_blank">jmoran@secureachsystems.com</a>>
wrote:<u></u><u></u></p>
<div>
<div>
<p class="MsoNormal">In
connecting to Nuance Speech Server’s MRCPv2 for text-to-speech (Nuance
Vocalizer) using mod_unimrcp, a single phone call which has several different
TTS phrases should only use up 1 single simultaneous license. However, I am
seeing a call use up 2,3,4 (etc) like it is not releasing the
connection/license.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal">I
looked this up and some forums have noted that Nuance “does things
wrong” and uniMRCP needs to send Nuance the “RTCP BYE” to
tell Nuance that it is done with the synthesizer speech. But I tried
that, and it still eats up the licenses. Within a couple of calls I’m out
of TTS ports and the calls fail (but they clear again after several more
seconds).<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><!-- enable/disable rtcp
support --></span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><param
name="rtcp" value="1"/></span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><!-- rtcp bye policies
(rtcp must be enabled first)</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> 0 -
disable rtcp bye</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> 1 -
send rtcp bye at the end of session</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> 2 -
send rtcp bye also at the end of each talkspurt (input)</span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New"">--></span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><param name="rtcp-bye"
value="2"/></span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""><param
name="rtcp-tx-interval" value="5000"/></span><u></u><u></u></p>
<p class="MsoNormal"><span style="font-size:10.0pt;font-family:"Courier New""> </span><u></u><u></u></p>
<p class="MsoNormal">Ideas?
Experience?<u></u><u></u></p>
<p class="MsoNormal"><span style="color:#888888">-Jason<u></u><u></u></span></p>
</div>
</div>
<p class="MsoNormal" style="margin-bottom:12.0pt"><br>
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<u></u><u></u></p>
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<div>
<p class="MsoNormal" style="margin-bottom:12.0pt">-- <br>
Michael S Collins<br>
Twitter: @mercutioviz<br>
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