Send PCAP with the RTSP signaling and RTP audio.  I&#39;m curious when the server starts streaming the audio.<br><br><br><div class="gmail_quote">On Fri, Feb 15, 2013 at 4:20 AM, Manuel Durán Aguete <span dir="ltr">&lt;<a href="mailto:manuel@aguete.org" target="_blank">manuel@aguete.org</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><span>Hello</span>, <div><br></div><div>I&#39;ve <span>tried</span> with different values for sleep, but the problem persists, in <a href="http://pastebin.freeswitch.org/20594" target="_blank">http://pastebin.freeswitch.org/20594</a> i&#39;ve posted the call log. </div>


<div>I&#39;ve enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in <a href="http://mod_unircp.cm" target="_blank">mod_unircp.cm</a>  the audio streams  generated can be downloaded from <a href="https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB" target="_blank">https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB</a>.</div>


<div><br></div><div>I&#39;ve changed the speak data to </div><div><br></div><div><br></div><div><div class="im"><div>   &lt;extension name=&quot;unimrcp&quot;&gt;</div><div>     &lt;condition field=&quot;destination_number&quot; expression=&quot;^5555$&quot;&gt;</div>


<div>       &lt;action application=&quot;answer&quot;/&gt;</div><div>       &lt;action application=&quot;set&quot; data=&quot;tts_engine=unimrcp&quot;/&gt;</div><div>       &lt;action application=&quot;set&quot; data=&quot;tts_voice=leonor&quot;/&gt;</div>


</div><div>       &lt;action application=&quot;sleep&quot; data=&quot;1000&quot;/&gt;</div><div>       &lt;action application=&quot;speak&quot; data=&quot;1 2 3 4 5 6 7 8 9 0&quot;/&gt;</div><div class="im"><div>       &lt;action application=&quot;sleep&quot; data=&quot;500&quot;/&gt;</div>


</div><div>       &lt;action application=&quot;hangup&quot;/&gt;</div><div>     &lt;/condition&gt;</div><div>   &lt;/extension&gt;</div><div><br></div><div>and only &quot;3 4 5 6 7 8 9 0&quot; is played in softphone. </div>


<div><br></div><div>Thank you for your help.</div><div><br></div><div>Regards.</div><div><br></div></div><div><br></div><div><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">


2013/2/14 Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>&gt;</span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


Try adding a sleep in your dialplan prior to the call to &quot;speak&quot; application. Start with 1000 ms and see if it makes any difference, then tweak the sleep time from there.<div><br></div><div>-MC<br><br><div class="gmail_quote">


<div><div>
On Thu, Feb 14, 2013 at 5:26 AM, Manuel Durán Aguete <span dir="ltr">&lt;<a href="mailto:manuel@aguete.org" target="_blank">manuel@aguete.org</a>&gt;</span> wrote:<br></div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">


<div><div>
<div dir="ltr">Hello,<div><br></div><div>I&#39;m a newbie testing freeswitch tts with loquendo mrcp1 server,</div><div><br></div><div>mrcp_profile:</div><div><br></div><div><div>&lt;include&gt;</div>

<div>  &lt;profile name=&quot;loquendo-mrcp1&quot; version=&quot;1&quot;&gt;</div><div>    &lt;param name=&quot;server-ip&quot; value=&quot;172.16.24.1&quot;/&gt;</div><div>    &lt;param name=&quot;server-port&quot; value=&quot;554&quot;/&gt;   </div>





<div>    &lt;param name=&quot;resource-location&quot; value=&quot;&quot;/&gt;</div><div>    &lt;param name=&quot;speechsynth&quot; value=&quot;synthesizer&quot;/&gt;</div><div>    &lt;param name=&quot;speechrecog&quot; value=&quot;recognizer&quot;/&gt;</div>





<div>    &lt;param name=&quot;rtp-ip&quot; value=&quot;$${local_ip_v4}&quot;/&gt;</div><div>    &lt;param name=&quot;rtp-port-min&quot; value=&quot;4000&quot;/&gt;</div><div>    &lt;param name=&quot;rtp-port-max&quot; value=&quot;5000&quot;/&gt;</div>





<div>    &lt;param name=&quot;codecs&quot; value=&quot;PCMU PCMA&quot;/&gt;</div><div>    &lt;synthparams&gt; </div><div>    &lt;/synthparams&gt;</div><div>    &lt;recogparams&gt;</div><div>    &lt;/recogparams&gt;</div>




<div>
  &lt;/profile&gt;</div><div>&lt;/include&gt;</div><div><br></div><div><br></div><div>Dialplan code:</div><div><div>   &lt;extension name=&quot;unimrcp&quot;&gt;</div><div>     &lt;condition field=&quot;destination_number&quot; expression=&quot;^5555$&quot;&gt;</div>





<div>       &lt;action application=&quot;answer&quot;/&gt;</div><div>       &lt;action application=&quot;set&quot; data=&quot;tts_engine=unimrcp&quot;/&gt;</div><div>       &lt;action application=&quot;set&quot; data=&quot;tts_voice=leonor&quot;/&gt;</div>





<div>       &lt;action application=&quot;speak&quot; data=&quot;Teclea o dime un número de 1 a 10&quot;/&gt;                     </div><div>       &lt;action application=&quot;sleep&quot; data=&quot;500&quot;/&gt;</div><div>





     &lt;/condition&gt;</div><div>   &lt;/extension&gt;  </div><div><br></div><div>I can hear all text with first word dropped. I&#39;ve captured the rtp traffic and the audio between FS &lt;-&gt; Loquendo contains the word. After apply the  fixed recommended in <a href="http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio" target="_blank">http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio</a> the issue persists.</div>





<div><br></div><div>Any idea, suggestion, etc ?</div><div><br></div><div>Thank you. </div><div><br></div><div><br></div></div></div></div>
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