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

[Christopher Rienzo]

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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