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

[David Brazier]

Hi Michael/Chris

Did you find out what the problem was?  We're seeing something similar.

Thanks in advance
David
On Mon Mar 14, Michel Habib <michelhabib at gmail.com>wrote:
>
> 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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