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.
-MC
On 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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