Audio dropped using unimrcp tts.
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="$${local_ip_v4}"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA"/> <synthparams> </synthparams> <recogparams> </recogparams> </profile> </include> Dialplan code: <extension name="unimrcp"> <condition field="destination_number" expression="^5555$"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp"/> <action application="set" data="tts_voice=leonor"/> <action application="speak" data="Teclea o dime un número de 1 a 10"/> <action application="sleep" data="500"/> </condition> </extension> I can hear all text with first word dropped. I've captured the rtp traffic and the audio between FS <-> Loquendo contains the word. After apply the fixed recommended in http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue persists. Any idea, suggestion, etc ? Thank you.
Try adding a sleep in your dialplan prior to the call to "speak" application. Start with 1000 ms and see if it makes any difference, then tweak the sleep time from there. -MC On Thu, Feb 14, 2013 at 5:26 AM, Manuel Durán Aguete <manuel@aguete.org>wrote:
Hello,
I'm a newbie testing freeswitch tts with loquendo mrcp1 server,
mrcp_profile:
<include> <profile name="loquendo-mrcp1" version="1"> <param name="server-ip" value="172.16.24.1"/> <param name="server-port" value="554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="synthesizer"/> <param name="speechrecog" value="recognizer"/> <param name="rtp-ip" value="$${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$"> <action application="answer"/> <action application="set" data="tts_engine=unimrcp"/> <action application="set" data="tts_voice=leonor"/> <action application="speak" data="Teclea o dime un número de 1 a 10"/> <action application="sleep" data="500"/> </condition> </extension>
I can hear all text with first word dropped. I've captured the rtp traffic and the audio between FS <-> Loquendo contains the word. After apply the fixed recommended in http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio the issue persists.
Any idea, suggestion, etc ?
Thank you.
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Hello, I've tried with different values for sleep, but the problem persists, in http://pastebin.freeswitch.org/20594 i've posted the call log. I've enabled MOD_UNIMRCP_DEBUG_AUDIO_QUEUE in mod_unircp.cm the audio streams generated can be downloaded from https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB. I've changed the speak data to <extension name="unimrcp"> <condition field="destination_number" expression="^5555$"> <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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hello, I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, added the log file too. https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB I've tested with branch 1.2 and with master, the pcap is from master code. Thank you for your help. Regards. 2013/2/15 Christopher Rienzo <cmrienzo@gmail.com>
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Sorry for the delay. I looked at the pcap and it confirmed my suspicion. The MRCP server doesn't reply with IN-PROGRESS until 1.35 sec of audio has already been transmitted. IN-PROGRESS tells mod_unimrcp that the request is has been accepted and is being processed. The module assumes the reply is immediately sent before the RTP is streamed. Chris On Mon, Feb 18, 2013 at 6:09 AM, Manuel Durán Aguete <manuel@aguete.org>wrote:
Hello,
I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, added the log file too.
https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB
I've tested with branch 1.2 and with master, the pcap is from master code.
Thank you for your help.
Regards.
2013/2/15 Christopher Rienzo <cmrienzo@gmail.com>
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.
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thank you Christopher, I'm going to review loquendo server for misconfgured settings or network issues. Regards. 2013/2/27 Christopher Rienzo <cmrienzo@gmail.com>
Sorry for the delay. I looked at the pcap and it confirmed my suspicion. The MRCP server doesn't reply with IN-PROGRESS until 1.35 sec of audio has already been transmitted.
IN-PROGRESS tells mod_unimrcp that the request is has been accepted and is being processed. The module assumes the reply is immediately sent before the RTP is streamed.
Chris
On Mon, Feb 18, 2013 at 6:09 AM, Manuel Durán Aguete <manuel@aguete.org>wrote:
Hello,
I've update the dropbox folder with the pcap and mod_unimrcp_rx/tx files, added the log file too.
https://www.dropbox.com/sh/tcgnktzzby0h2ns/v7ViiUyObB
I've tested with branch 1.2 and with master, the pcap is from master code.
Thank you for your help.
Regards.
2013/2/15 Christopher Rienzo <cmrienzo@gmail.com>
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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_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (3)
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Christopher Rienzo
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Manuel Durán Aguete
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Michael Collins