Send PCAP with the RTSP signaling and RTP audio. I'm curious when the server starts streaming the audio.On Fri, Feb 15, 2013 at 4:20 AM, Manuel Durán Aguete <manuel@aguete.org> wrote:
Hello,I've tried with different values for sleep, but the problem persists, in http://pastebin.freeswitch.org/20594 i've posted the call log.I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio streams generated can be downloaded from https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB.I've changed the speak data to<extension name="unimrcp"><condition field="destination_number" expression="^5555$"><action application="answer"/><action application="set" data="tts_engine=unimrcp"/><action application="set" data="tts_voice=leonor"/><action application="sleep" data="1000"/><action application="speak" data="1 2 3 4 5 6 7 8 9 0"/><action application="sleep" data="500"/><action application="hangup"/></condition></extension>and only "3 4 5 6 7 8 9 0" is played in softphone.Thank you for your help.Regards.2013/2/14 Michael Collins <msc@freeswitch.org>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.-MCOn Thu, Feb 14, 2013 at 5:26 AM, Manuel Durán Aguete <manuel@aguete.org> wrote:_________________________________________________________________________Hello,I'm a newbie testing freeswitch tts with loquendo mrcp1 server,mrcp_profile:<include><profile name="loquendo-mrcp1" version="1"><param name="server-ip" value="172.16.24.1"/><param name="server-port" value="554"/><param name="resource-location" value=""/><param name="speechsynth" value="synthesizer"/><param name="speechrecog" value="recognizer"/><param name="rtp-ip" value="$${local_ip_v4}"/><param name="rtp-port-min" value="4000"/><param name="rtp-port-max" value="5000"/><param name="codecs" value="PCMU PCMA"/><synthparams></synthparams><recogparams></recogparams></profile></include>Dialplan code:<extension name="unimrcp"><condition field="destination_number" expression="^5555$"><action application="answer"/><action application="set" data="tts_engine=unimrcp"/><action application="set" data="tts_voice=leonor"/><action application="speak" data="Teclea o dime un número de 1 a 10"/><action application="sleep" data="500"/></condition></extension>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 http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue persists.Any idea, suggestion, etc ?Thank you.
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