Audio problems with unimrcp and Vestec ASR
I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" What does work: * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. What does NOT work: * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout) Here are my configs: unimrcp.conf.xml <configuration name="unimrcp.conf" description="UniMRCP Client"> <settings> <param name="default-tts-profile" value="unimrcpserver-mrcp1"/> <param name="default-asr-profile" value="unimrcpserver-mrcp1"/> <param name="log-level" value="DEBUG"/> <param name="enable-profile-events" value="false"/> <param name="max-connection-count" value="100"/> <param name="offer-new-connection" value="1"/> </settings> <profiles> <X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/> </profiles> </configuration> unimrcpserver-mrcp-v1.xml <include> <!-- UniMRCP Server MRCPv1 --> <profile name="unimrcpserver-mrcp1" version="1"> <param name="server-ip" value="192.168.178.180"/> <param name="server-port" value="1554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="speechsynthesizer"/> <param name="speechrecog" value="speechrecognizer"/> <param name="rtp-ip" value="auto"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA L16/96/8000"/> <synthparams> </synthparams> <!-- Add any default MRCP params for RECOGNIZE requests here --> <recogparams> <!--param name="start-input-timers" value="false"/--> </recogparams> </profile> </include> See logs here http://pastebin.freeswitch.org/18570 Any help is appreciated. Best egards Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
What codec is used on the call leg? I tried some ASR (UniMRCP to Nuance) once, and I noticed that the audio passed to Nuance seemed to fail when using HD audio, when I did the same test with standard PCMA it worked. I never had the time to investigate any further though, since this was onle for a test, and nothing I needed to implement yet. /Peter ________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Peter Steinbach [lists@telefaks.de] Skickat: den 4 mars 2012 14:57 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR I have an Audio problem with unimrcp and Vestec ASR with mrcp v1. This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem" What does work: * MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully. What does NOT work: * RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout) Here are my configs: unimrcp.conf.xml <configuration name="unimrcp.conf" description="UniMRCP Client"> <settings> <param name="default-tts-profile" value="unimrcpserver-mrcp1"/> <param name="default-asr-profile" value="unimrcpserver-mrcp1"/> <param name="log-level" value="DEBUG"/> <param name="enable-profile-events" value="false"/> <param name="max-connection-count" value="100"/> <param name="offer-new-connection" value="1"/> </settings> <profiles> <X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/> </profiles> </configuration> unimrcpserver-mrcp-v1.xml <include> <!-- UniMRCP Server MRCPv1 --> <profile name="unimrcpserver-mrcp1" version="1"> <param name="server-ip" value="192.168.178.180"/> <param name="server-port" value="1554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="speechsynthesizer"/> <param name="speechrecog" value="speechrecognizer"/> <param name="rtp-ip" value="auto"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA L16/96/8000"/> <synthparams> </synthparams> <!-- Add any default MRCP params for RECOGNIZE requests here --> <recogparams> <!--param name="start-input-timers" value="false"/--> </recogparams> </profile> </include> See logs here http://pastebin.freeswitch.org/18570 Any help is appreciated. Best egards Peter -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de<http://www.telefaks.de> !DSPAM:4f5373f932761937219840!
Hello Peter, thanks for the hint. It was using L16 codec, so I changed the avaliable codecs for the mrcp profile, so that it only accepts PCMA. Now it negociates PCMA, see FS log: 2012-03-04 18:12:40.266537 [INFO] rtsp_client.c:929 () Receive RTSP Stream 192.168.178.221:54797 <-> 192.168.178.180:1554 [324 bytes] RTSP/1.0 200 OK CSeq: 1 Transport: RTP/AVP;unicast;client_port=4008-4009;server_port=5022-5023 Session: dafbbc3e7efb4474 Content-Type: application/sdp Content-Length: 145 v=0 o=UniMRCPServer 0 0 IN IP4 192.168.178.180 s=- c=IN IP4 192.168.178.180 t=0 0 m=audio 5022 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=recvonly But the behaviour is the same. The RTP stream I grepped on the network seems always to be empty (silence). All UDP RTP Packets sent from FS look like this (hex values): 80:08:00:08:00:00:05:00:f9:53:b5:e6:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5 Best regards Peter Am 04.03.2012 15:17, schrieb Peter Olsson:
What codec is used on the call leg? I tried some ASR (UniMRCP to Nuance) once, and I noticed that the audio passed to Nuance seemed to fail when using HD audio, when I did the same test with standard PCMA it worked. I never had the time to investigate any further though, since this was onle for a test, and nothing I needed to implement yet.
/Peter
________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Peter Steinbach [lists@telefaks.de] Skickat: den 4 mars 2012 14:57 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR
I have an Audio problem with unimrcp and Vestec ASR with mrcp v1.
This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem"
What does work:
* MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully.
What does NOT work:
* RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout)
Here are my configs: unimrcp.conf.xml <configuration name="unimrcp.conf" description="UniMRCP Client"> <settings> <param name="default-tts-profile" value="unimrcpserver-mrcp1"/> <param name="default-asr-profile" value="unimrcpserver-mrcp1"/> <param name="log-level" value="DEBUG"/> <param name="enable-profile-events" value="false"/>
<param name="max-connection-count" value="100"/> <param name="offer-new-connection" value="1"/> </settings> <profiles> <X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/> </profiles> </configuration>
unimrcpserver-mrcp-v1.xml <include> <!-- UniMRCP Server MRCPv1 --> <profile name="unimrcpserver-mrcp1" version="1"> <param name="server-ip" value="192.168.178.180"/> <param name="server-port" value="1554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="speechsynthesizer"/> <param name="speechrecog" value="speechrecognizer"/> <param name="rtp-ip" value="auto"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA L16/96/8000"/> <synthparams> </synthparams> <!-- Add any default MRCP params for RECOGNIZE requests here --> <recogparams> <!--param name="start-input-timers" value="false"/--> </recogparams> </profile> </include>
See logs here http://pastebin.freeswitch.org/18570
Any help is appreciated.
Best egards Peter
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de<http://www.telefaks.de>
!DSPAM:4f5373f932761937219840!
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
Just for the records... after extensive testing with Vestec support, we were still at the point that Freeswitch was sending empty RTP packets to the MRCP server. After updating Freeswitch today to newest GIT (our previous GIT version was from 29-Jan-2012), recognition via MRCP worked successfully. Thanks to all who supported on this issue. Best regards Peter Am 04.03.2012 18:25, schrieb Peter Steinbach:
Hello Peter,
thanks for the hint. It was using L16 codec, so I changed the avaliable codecs for the mrcp profile, so that it only accepts PCMA. Now it negociates PCMA, see FS log:
2012-03-04 18:12:40.266537 [INFO] rtsp_client.c:929 () Receive RTSP Stream 192.168.178.221:54797 <-> 192.168.178.180:1554 [324 bytes] RTSP/1.0 200 OK CSeq: 1 Transport: RTP/AVP;unicast;client_port=4008-4009;server_port=5022-5023 Session: dafbbc3e7efb4474 Content-Type: application/sdp Content-Length: 145 v=0 o=UniMRCPServer 0 0 IN IP4 192.168.178.180 s=- c=IN IP4 192.168.178.180 t=0 0 m=audio 5022 RTP/AVP 8 a=rtpmap:8 PCMA/8000 a=recvonly
But the behaviour is the same. The RTP stream I grepped on the network seems always to be empty (silence). All UDP RTP Packets sent from FS look like this (hex values): 80:08:00:08:00:00:05:00:f9:53:b5:e6:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5:d5
Best regards Peter
Am 04.03.2012 15:17, schrieb Peter Olsson:
What codec is used on the call leg? I tried some ASR (UniMRCP to Nuance) once, and I noticed that the audio passed to Nuance seemed to fail when using HD audio, when I did the same test with standard PCMA it worked. I never had the time to investigate any further though, since this was onle for a test, and nothing I needed to implement yet.
/Peter
________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Peter Steinbach [lists@telefaks.de] Skickat: den 4 mars 2012 14:57 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] Audio problems with unimrcp and Vestec ASR
I have an Audio problem with unimrcp and Vestec ASR with mrcp v1.
This seems to be a similar problem to thread: "ASR from Freeswitch to MS Speech Server [using MRCP Connector] - Audio Problem"
What does work:
* MRCP command are passed successsfull in both ways on port 1554 (vestec mrcp server) * Grammar is accepted and recognition is started * I can see RTP stream (wireshark) from FS port 400x to the mrcp server * Result (002 no-input-timeout) is sent back to Freeeswicth and is processed by FS successfully.
What does NOT work:
* RTP stream is always empty (silence), i grepped this on the network with ngrep, RTP data is empty, there is no change in the audio data when I speak. * so voice recognition will always time out (002 no-input-timeout)
Here are my configs: unimrcp.conf.xml <configuration name="unimrcp.conf" description="UniMRCP Client"> <settings> <param name="default-tts-profile" value="unimrcpserver-mrcp1"/> <param name="default-asr-profile" value="unimrcpserver-mrcp1"/> <param name="log-level" value="DEBUG"/> <param name="enable-profile-events" value="false"/>
<param name="max-connection-count" value="100"/> <param name="offer-new-connection" value="1"/> </settings> <profiles> <X-PRE-PROCESS cmd="include" data="../mrcp_profiles/*.xml"/> </profiles> </configuration>
unimrcpserver-mrcp-v1.xml <include> <!-- UniMRCP Server MRCPv1 --> <profile name="unimrcpserver-mrcp1" version="1"> <param name="server-ip" value="192.168.178.180"/> <param name="server-port" value="1554"/> <param name="resource-location" value=""/> <param name="speechsynth" value="speechsynthesizer"/> <param name="speechrecog" value="speechrecognizer"/> <param name="rtp-ip" value="auto"/> <param name="rtp-port-min" value="4000"/> <param name="rtp-port-max" value="5000"/> <param name="codecs" value="PCMU PCMA L16/96/8000"/> <synthparams> </synthparams> <!-- Add any default MRCP params for RECOGNIZE requests here --> <recogparams> <!--param name="start-input-timers" value="false"/--> </recogparams> </profile> </include>
See logs here http://pastebin.freeswitch.org/18570
Any help is appreciated.
Best egards Peter
-- With kind regards Peter Steinbach
Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de<http://www.telefaks.de>
!DSPAM:4f5373f932761937219840!
_________________________________________________________________________
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
-- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de
participants (2)
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Peter Olsson
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Peter Steinbach