[Freeswitch-users] ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem

[Michel Habib]

Sorry for the delay and thanks Chris,
Please find attached logs: http://www.mediafire.com/?8thg9ey7kd26882
sip client ip 192.168.1.202
freeswitch ip 192.168.1.35
mrcp connector/speech server ip 192.168.1.32
i have attached 3 logs - freeswitch log file, wireshark on freeswitch
server, wireshark on mrcp connector/speech server.
my script simply plays back audio from wave [basic freeswitch function],
then plays back audio from TTS [which cannot be heard]

Thank you,
Michel.

---------- Forwarded message ----------
> From: Christopher Rienzo <cmrienzo at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 9 Mar 2011 22:25:34 -0500
> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
> [using MRCP Connector] - Audio Problem
> I still would like to see the logs for your call.
>
>
> On Tue, Mar 8, 2011 at 7:18 AM, Michel Habib <michelhabib at gmail.com>wrote:
>
>> Yes, I get the Audio from FS in regular calls - I already disabled all
>> possible firewalls - all 3 machines [softphone, freeswitch, Speech Server
>> (and mrcp connector) ] are on a switch.
>> 192.168.5.107 is the freeswitch server
>> 192.168.5.110 is the MRCP-Connector/Speech Server 2007 server
>> I made too many iterations on the configuration below:
>>
>> <include>
>>   <profile name="mrcp-connector" version="2">
>>     <param name="client-ip" value="192.168.5.107"/>
>>     <param name="client-port" value="5090"/>
>>     <param name="server-ip" value="192.168.5.110"/>
>>     <param name="server-port" value="5070"/>
>>     <!--param name="force-destination" value="1"/-->
>>     <param name="sip-transport" value="udp"/>
>>     <!--param name="ua-name" value="FreeSWITCH"/-->
>>     <!--param name="sdp-origin" value="FreeSWITCH"/-->
>>     <param name="rtp-ext-ip" value="192.168.5.107"/>
>>     <param name="rtp-ip" value="192.168.5.107"/>
>>     <param name="rtp-port-min" value="4000"/>
>>     <param name="rtp-port-max" value="5000"/>
>>     <!-- enable/disable rtcp support -->
>>     <param name="rtcp" value="1"/>
>>     <!-- rtcp bye policies (rtcp must be enabled first)
>>              0 - disable rtcp bye
>>              1 - send rtcp bye at the end of session
>>              2 - send rtcp bye also at the end of each talkspurt (input)
>>     -->
>>     <param name="rtcp-bye" value="2"/>
>>     <!-- rtcp transmission interval in msec (set 0 to disable) -->
>>     <param name="rtcp-tx-interval" value="5000"/>
>>     <!-- period (timeout) to check for new rtcp messages in msec (set 0 to
>> disable) -->
>>     <param name="rtcp-rx-resolution" value="1000"/>
>>
>>     <!--param name="playout-delay" value="50"/-->
>>     <!--param name="max-playout-delay" value="200"/-->
>>     <!--param name="ptime" value="20"/-->
>>     <param name="codecs" value="PCMU PCMA L16/96/8000"/>
>>
>>     <!-- Add any default MRCP params for SPEAK requests here -->
>>     <synthparams>
>>     </synthparams>
>>
>>     <!-- Add any default MRCP params for RECOGNIZE requests here -->
>>     <recogparams>
>>       <!--param name="start-input-timers" value="false"/-->
>>     </recogparams>
>>   </profile>
>> </include>
>>
>>
>>
>> ---------- Forwarded message ----------
>>
>>> From: Christopher Rienzo <cmrienzo at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 7 Mar 2011 09:32:51 -0500
>>> Subject: Re: [Freeswitch-users] ASR from Freeswitch to MS Speech Server
>>> [using MRCP Connector] - Audio Problem
>>> Do you get audio between FS and your SIP client when not using ASR/TTS?
>>>
>>> Show me the MRCP profile configuration and your FreeSWITCH logs during
>>> the call.
>>>
>>>
>>>
>>> On Mon, Mar 7, 2011 at 4:55 AM, Michel Habib <michelhabib at gmail.com>wrote:
>>>
>>>> Hello All,
>>>> I have MS OCS Speech Server 2007 [working correctly, as i can make SIP
>>>> calls and use its ASR and TTS Services successfully]
>>>> I am also using MRCP Connector from AumTech - which allows me to use ASR
>>>> and TTS Services through an MRCP Client .
>>>> Now, i am using Freeswitch mod unimrcp to use ASR and TTS.
>>>>
>>>> for TTS, I can successfully make the call, the Audio RTP of the TTS
>>>> voice is transferred succesfully from Speech Server [through MRCP Connector]
>>>> back to the Freeswitch Server.
>>>> However, Freeswitch is not sending back the Audio RTP to the SIP client.
>>>>
>>>> for ASR, I can successfully define the grammar and start recognition,
>>>> but the audio RTP sent to speech server [through MRCP Connector] is silent
>>>> [empty].
>>>>
>>>> I am suspecting something is wrong with the RTP Configuration - can you
>>>> help me?
>>>>
>>>> Let me now if you need any specific logs/scripts/configuration?
>>>>
>>>> Thank you,
>>>> Michel.
>>>>
>>>
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