Configuration Guide: Unimrcp On Asterisk
Configuration Guide: Unimrcp On Asterisk
Configuration Guide: Unimrcp On Asterisk
Configuration Guide
UniMRCP on Asterisk
version 2.0.0
Summary
Summary 1
1. Introduction 2
1.1 What is UniMrcp? 2
1.2 What subjects are covered in this guide? 2
1.3 What subjects are not covered in this guide? 2
1.4 Legal notice 2
2. Environment preparation 3
4. Troubleshooting 7
4.1 I can’t listen anything 7
4.2 Error loading libsofia library 7
5. Useful links 8
1
1. Introduction
2
2. Environment preparation
This guide covers how to configure the UniMRCP module for Asterisk with
Verbio MRCP Server. But in order to continue with the configuration your
environment must be prepared. Check whether next steps are met:
1. Install and configure Verbio Voxserver (follow installing and
configuring verbio from Verbio’s installation guide)
2. Install and configure Verbio MRCP ( same as point 1)
3. Install Asterisk ( Asterisk installation - wiki)
4. Install UniMRCP Components (UniMRCP components)
a. UniMRCP Dependencies
b. UniMRCP
c. UniMRCP for Asterisk
3
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rtp-port-max = 5000
; === Jitter buffer settings ===
playout-delay = 50
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
; Profile for Verbio’s MRCP Server version 1
[verbio-mrcpv2]
; MRCP version
version = 2
; === SIP settings ===
; Must be set to Verbio’s MRCP Server information (/opt/verbio/config/software-verbio-mrcp default
5060)
server-ip = TO-BE-DEFINED
server-port = 5060
force-destination = 1
; === SIP agent ===
; same IP where Asterisk is running that will be used to connect to MRCP Server
client-ip = TO-BE-DEFINED
; SIP port that will be used and type (tcp, upd)
client-port = 8060
sip-transport = udp
; === RTP Factory ===
; same IP where Asterisk is running that will be used to connect to MRCP Server
rtp-ip = TO-BE-DEFINED
; RTP port range that will be used
rtp-port-min = 4000
rtp-port-max = 5000
; === Jitter buffer settings ===
playout-delay = 50
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
5
rtcp-tx-interval = 5000
rtcp-rx-resolution = 1000
These definitions can be copied and pasted directly to /etc/asterisk/mrcp.conf but
check the values of:
✓ s erver-ip
✓ s erver-port
✓ r tp-ip
✓ r tp-port-min and rtp-port-max
✓ c lient-ip
✓ c lient-port
✓ s ip-transport
Also take into account that Asterisk uses port 5060 for SIP by default that
means that if Verbio’s MRCP Server and Asterisk are in the same machine
one should have a different port configured.
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4. Troubleshooting
7
5. Useful links
1. UniMRCP Plugin Manual for Asterisk
2. RFC 4663 - MRCP (version 1)
3. RFC 6787 - MRCPv2
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