;
; General settings
;
[general]
; Default ASR and TTS profiles.
default-asr-profile = speech-ali-mrcp2
; UniMRCP logging level to appear in Asterisk logs.  Options are:
; EMERGENCY|ALERT|CRITICAL|ERROR|WARNING|NOTICE|INFO|DEBUG -->
log-level = DEBUG
max-connection-count = 200
max-shared-count = 200
offer-new-connection = 1
; rx-buffer-size = 1024
; tx-buffer-size = 1024
request-timeout = 8000
; speech-channel-timeout = 30000


;
; Profile for Ali Speech Server [MRCPv2]
;
[speech-ali-mrcp2]
; MRCP version.
version = 2

; === SIP settings ===
; Must be set to the IP address of the MRCP server.
server-ip = 175.178.41.235
; SIP port on the MRCP server.
server-port = 8170
; server-username = test
force-destination = 1

; === SIP agent ===
; Must be set to the IP address of the MRCP client.
client-ip = 192.168.19.140
;client-ext-ip = auto
; SIP port on the MRCP client.
client-port = 5093
; SIP transport either UDP or TCP.
sip-transport = udp
; ua-name = Asterisk
; sdp-origin = Asterisk
; sip-t1 = 500
; sip-t2 = 4000
; sip-t4 = 4000
; sip-t1x64 = 32000
; sip-timer-c = 185000

; === RTP factory ===
; Must be set to the IP address of the MRCP client.
rtp-ip = 192.168.19.140
;rtp-ext-ip = auto
; RTP port range on the MRCP client.
rtp-port-min = 10000
rtp-port-max = 20000

; === Jitter buffer settings ===
playout-delay = 50
; min-playout-delay = 20
max-playout-delay = 200

; === RTP settings ===
ptime = 20
codecs = PCMU PCMA L16/96/8000 telephone-event/101/8000

; === RTCP settings ===
rtcp = 1
rtcp-bye = 2
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000