No RPT when test against IBM WVS 5.1.3

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I'm new to UniMRCP, just tried asterisk-unimrcp 0.3 against IBM WVS 5.1.3.
the following is my dialplan:
exten => 1001,1,Answer
exten => 1001,n,MRCPSynth(Hello 
world!,p=default&i=any&f=/tmp/synth.raw&l=en-US&v=Allison&g=female)
exten => 1001,n,Hangup

Got "Failed to Find Free RTP Port" error, detailed error please see attached 
asterisk log.

I also use ngrep took some packet track (attached ngrep.log please see the 
detailed trace), but the ngrep trace looks all right.

Any suggestion about using asterisk-unimrcp to access IBM WVS, Does any special 
cofigurstion do I missing? the following is my mrcp.conf

[general]
default-asr-profile = speech-nuance5-mrcp1
default-tts-profile = speech-nuance5-mrcp1
; UniMRCP logging level to appear in Asterisk logs.  Options are:
; EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG -->
log-level = DEBUG
max-connection-count = 100
offer-new-connection = 1
; rx-buffer-size = 1024
; tx-buffer-size = 1024
; request-timeout = 60

[speech-nuance5-mrcp1]
; +++ MRCP settings +++
version = 1
;
; +++ RTSP +++
; === RSTP settings ===
server-ip = 192.168.0.15
server-port = 554
force-destination = 1
resource-location = media
speechsynth = synthesizer
speechrecog = recognizer
;
; +++ RTP +++
; === RTP factory ===
; rtp-ip = 0.0.0.0
rtp-ip = 192.168.0.15
; rtp-ext-ip = auto
rtp-port-min = 4000
rtp-port-max = 5000
; === RTP settings ===
; --- Jitter buffer settings ---
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200
; --- RTP settings ---
ptime = 20
codecs = PCMU PCMA L16/96/8000
; --- RTCP settings ---
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000

Thanks
Peter

Original issue reported on code.google.com by guibin...@gmail.com on 10 Oct 2010 at 6:26

Attachments:

• ngrep.log
• asterisk.log

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[deleted comment]

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[deleted comment]

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Hi Peter,

Sorry for the late response. This indeed looks like a small configuration 
issue. As opposite to the server-ip address, the rtp-ip should be a local ip 
address to bind to, and thus be used for RTP streaming. The rest should be OK.

Original comment by achalo...@gmail.com on 21 Dec 2010 at 9:44

• Added labels: Component-Asterisk, OpSys-Linux, Priority-Medium, Type-Task

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Original comment by achalo...@gmail.com on 15 Jul 2011 at 8:17

• Changed state: Verified