<div dir="ltr">Hello, <div><br></div><div style>I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, added the log file too.</div><div style><br></div><div><a href="https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB">https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB</a><br>
</div><div><br></div><div style>I've tested with branch 1.2 and with master, the pcap is from master code.</div><div><br></div><div style>Thank you for your help.</div><div style><br></div><div style>Regards.</div><div>
<br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">2013/2/15 Christopher Rienzo <span dir="ltr"><<a href="mailto:cmrienzo@gmail.com" target="_blank">cmrienzo@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send PCAP with the RTSP signaling and RTP audio. I'm curious when the server starts streaming the audio.<div class="HOEnZb">
<div class="h5"><br><br><br><div class="gmail_quote">On Fri, Feb 15, 2013 at 4:20 AM, Manuel Durán Aguete <span dir="ltr"><<a href="mailto:manuel@aguete.org" target="_blank">manuel@aguete.org</a>></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'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've posted the call log. </div>
<div>I'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've changed the speak data to </div><div><br></div><div><br></div><div><div><div> <extension name="unimrcp"></div><div> <condition field="destination_number" expression="^5555$"></div>
<div> <action application="answer"/></div><div> <action application="set" data="tts_engine=unimrcp"/></div><div> <action application="set" data="tts_voice=leonor"/></div>
</div><div> <action application="sleep" data="1000"/></div><div> <action application="speak" data="1 2 3 4 5 6 7 8 9 0"/></div><div><div> <action application="sleep" data="500"/></div>
</div><div> <action application="hangup"/></div><div> </condition></div><div> </extension></div><div><br></div><div>and only "3 4 5 6 7 8 9 0" 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><div><div class="gmail_extra"><br><br><div class="gmail_quote">
2013/2/14 Michael Collins <span dir="ltr"><<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>></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 "speak" 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"><<a href="mailto:manuel@aguete.org" target="_blank">manuel@aguete.org</a>></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'm a newbie testing freeswitch tts with loquendo mrcp1 server,</div><div><br></div><div>mrcp_profile:</div><div><br></div><div><div><include></div>
<div> <profile name="loquendo-mrcp1" version="1"></div><div> <param name="server-ip" value="172.16.24.1"/></div><div> <param name="server-port" value="554"/> </div>
<div> <param name="resource-location" value=""/></div><div> <param name="speechsynth" value="synthesizer"/></div><div> <param name="speechrecog" value="recognizer"/></div>
<div> <param name="rtp-ip" value="$${local_ip_v4}"/></div><div> <param name="rtp-port-min" value="4000"/></div><div> <param name="rtp-port-max" value="5000"/></div>
<div> <param name="codecs" value="PCMU PCMA"/></div><div> <synthparams> </div><div> </synthparams></div><div> <recogparams></div><div> </recogparams></div>
<div>
</profile></div><div></include></div><div><br></div><div><br></div><div>Dialplan code:</div><div><div> <extension name="unimrcp"></div><div> <condition field="destination_number" expression="^5555$"></div>
<div> <action application="answer"/></div><div> <action application="set" data="tts_engine=unimrcp"/></div><div> <action application="set" data="tts_voice=leonor"/></div>
<div> <action application="speak" data="Teclea o dime un número de 1 a 10"/> </div><div> <action application="sleep" data="500"/></div><div>
</condition></div><div> </extension> </div><div><br></div><div>I can hear all text with first word dropped. I've captured the rtp traffic and the audio between FS <-> 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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