Overview

The PBX supports three types of SIP trunk groups: SIP AT0, SIP PRA, and SIP IP. A SIP IP trunk uses SIP based on IP. A SIP IP trunk group implements voice communication between PBX users and inter-office users.

The PBX at one end of a SIP IP trunk does not need to register with the device at the other end because the relationship between the two ends is equivalence relationship. Unlike a circuit trunk that defines a physical channel, a SIP IP trunk defines a logical channel and solves authentication and addressing problems between local and remote offices. When you configure a SIP IP trunk for a PBX, the remote end must support the SIP IP trunk, as shown in Figure 1-1

Figure 1-1 Networking of SIP IP trunk groups

 Configuration Notes

l   Table 1-4 describes matching AR models.

Table 1-1 Matching AR models

l   The voice functions are controlled by the license. By default, voice functions are disabled on a newly purchased device. To use the voice functions, apply for and purchase the license from Huawei local office.

l   In this example, the commands on the Cisco device are recommended ones. The product version is Cisco IOS Software, C2900 Software (C2900-UNIVERSALK9-M), Version 15.1(4)M1, RELEASE SOFTWARE (fc1). For details, visit http://www.cisco.com/techsupport.

Networking Requirements

In Figure 1-2, the AR connects to the phone of user 6201 and Cisco 2900 connects to the phone of user 6101. A SIP IP trunk is established between the AR and Cisco 2900 to implement communication between user 6201 and user 6101.

Figure 1-2 Connecting the AR to the Cisco 2900 through a SIP IP trunk

Data Plan

Before the configuration, plan data according to Table 1-1and Table 1-2. The data in this document is used for reference only.

Table 1-1 Data preparation for connecting the AR to the Cisco 2900

 Table 1-2 Plan of the SIP IP trunk

Configuring the AR

Configuration Roadmap

The configuration roadmap is as follows:

1.      Configure interface addresses and address pools.

2.      Configure a voice user.

3.      Configure a call route and a SIP IP trunk group.

4.      Configure intra-office and inter-office call prefixes.

Procedure

                      Step 1    Configure interface addresses and address pools.

<Huawei> interface gigabitethernet 0/0/0
[Huawei-GigabitEthernet0/0/0] ip address 10.1.1.1 24
[Huawei-GigabitEthernet0/0/0] quit
[Huawei] voice
[Huawei-voice] voip-address media interface gigabitethernet 0/0/0 10.1.1.1
[Huawei-voice] voip-address signalling interface gigabitethernet 0/0/0 10.1.1.1
[Huawei-voice] quit

                      Step 2    Configure a voice user.

[Huawei] voice
[Huawei-voice] pbxuser 6201 pots
[Huawei-voice-pbxuser-6201] port 2/0/0
[Huawei-voice-pbxuser-6201] telno 6201
[Huawei-voice-pbxuser-6201] quit

                      Step 3    Configure a call route and a SIP IP trunk group.

# Configure a call route sip.

[Huawei-voice] callroute sip
[Huawei-voice-callroute-sip] quit

# Create and configure a trunk group sip.

[Huawei-voice] trunk-group sip sip no-register
[Huawei-voice-trunkgroup-sip] callroute sip
[Huawei-voice-trunkgroup-sip] signalling-address ip 10.1.1.1 port 5062
[Huawei-voice-trunkgroup-sip] media-ip 10.1.1.1
[Huawei-voice-trunkgroup-sip] peer-address static 10.1.1.2 5060
[Huawei-voice-trunkgroup-sip] register-uri huawei.com
[Huawei-voice-trunkgroup-sip] home-domain huawei.com
[Huawei-voice-trunkgroup-sip] number-parameter 7 1
[Huawei-voice-trunkgroup-sip] number-parameter 10 1
[Huawei-voice-trunkgroup-sip] number-parameter 27 0
[Huawei-voice-trunkgroup-sip] number-parameter 53 0
[Huawei-voice-trunkgroup-sip] reset
Are you sure to continue?[Y/N]:y
 Note: Trunkgroup reset succeeds.
[Huawei-voice-trunkgroup-sip] quit

                      Step 4    Configure intra-office and inter-office call prefixes.

# Configure an intra-office call prefix. Create a call prefix profile 62, set the call prefix to 62, specify the basic service, configure the intra-office call, and set the maximum and minimum lengths of a number that can be parsed to 5 and 2 respectively.

[Huawei-voice] callprefix 62
[Huawei-voice-callprefix-62] prefix 62
[Huawei-voice-callprefix-62] call-type category basic-service attribute 0
[Huawei-voice-callprefix-62] digit-length 2 5
[Huawei-voice-callprefix-62] quit

# Configure an inter-office call prefix. Create a call prefix profile 61, set the call prefix to 61, specify the basic service, configure the local call, set the maximum and minimum lengths of a number that can be parsed to 5 and 2 respectively, and bind the call prefix profile to the call route.

[Huawei-voice] callprefix 61
[Huawei-voice-callprefix-62] prefix 61
[Huawei-voice-callprefix-62] call-type category basic-service attribute 1
[Huawei-voice-callprefix-62] digit-length 2 5
[Huawei-voice-callprefix-62] callroute sip
[Huawei-voice-callprefix-62] quit

                      Step 5    Save the configuration.

[Huawei-voice] save

----End

 Configuring the Cisco 2900

Configuration Roadmap

The configuration roadmap is as follows:

1.      Configure the Telnet mode for logging in to the Cisco 2900.

2.      Configure interface addresses.

3.      Configure VoIP parameters and enable SIP to SIP.

4.      Configure a SIP trunk to the AR.

5.      Configure a voice user on the Cisco 2900.

Procedure

                      Step 1    Configure the Telnet mode for logging in to the Cisco 2900.

<Huawei> telnet 10.1.1.2

                      Step 2    Configure interface addresses.

Router# configure
Configuring from terminal,memory,or network [terminal]?
Enter configuration commands,one per line.  End with CNTL/Z
Router(config)# interface GigabitEthernet0/1
Router(config-if)# ip address 10.1.1.2 255.255.255.0
Router(config-if)# exit

                      Step 3    Configure VoIP parameters, enable SIP to SIP, and bind SIP signaling and media to GigabitEthernet0/1.

Router(config)# voice service voip
Router(conf-voi-serv)# allow-connections sip to sip
Router(conf-serv-sip)# bind control source-interface GigabitEthernet0/1
Router(conf-serv-sip)# bind media source-interface GigabitEthernet0/1
Router(conf-serv-sip)# registrar server
Router(conf-serv-sip)# exit
Router(conf-voi-serv)# exit
Router(config)# voice class codec 1
Router(config-class)# codec preference 1 g711ulaw
Router(config-class)# codec preference 2 g711alaw
Router(config-class)# codec preference 3 g723r63
Router(config-class)# codec preference 4 g729r8
Router(config-class)# codec preference 5 g729br8

                      Step 4    Configure a SIP trunk to the AR.

Router(config-class)# dial-peer voice 62 voip
Router(config-dial-peer)#destination-pattern 6201
Router(config-dial-peer)# session protocol sipv2
Router(config-dial-peer)# session target ipv4:10.1.1.1:5062
Router(config-dial-peer)# incoming called-number 6201
Router(config-dial-peer)# codec g711alaw
Router(config-dial-peer)# exit

                      Step 5    Configure a voice user on the Cisco 2900.

Router(config)# dial-peer voice 6101 pots
Router(config-dial-peer)# destination-pattern 6101
Router(config-dial-peer)# port 0/0/0

----End

Verification

1.      Run the display voice trunk-group sip trunk-circuit state brief command. You can see that the link status of the SIP trunk on the AR is Idle.

<Huawei> system-view
[Huawei] voice
[Huawei-voice] display voice trunk-group sip trunk-circuit state brief
KC under trunk group state: sip
 -------------------------------
 TKC No  State
 2412    Idle
 2413    Idle
 2414    Idle
 2415    Idle
 2416    Idle
 2417    Idle
 2418    Idle
 2419    Idle
 2420    Idle
 2421    Idle
 2422    Idle
 2423    Idle
 2424    Idle
 2425    Idle
 2426    Idle
 2427    Idle
 2428    Idle
 2429    Idle
 2430    Idle
 2431    Idle
 2432    Idle
 2433    Idle
 2434    Idle
 2435    Idle
 2436    Idle
 2437    Idle
 2438    Idle
 2439    Idle
 2440    Idle
 2441    Idle

2.      Check communication information on the Cisco 2900.

 
Router# show voice call status
CallID     CID  ccVdb      Port        Slot/DSP:Ch  Called #   Codec    MLPP Dial-peers
0x356D     7BC  0x29798E68 0/0/0            0/1:1  *6101       g711alaw  62/6101
1 active call found
Router# show voice call summary
PORT           CODEC     VAD VTSP STATE            VPM STATE
============== ========= === ==================== ======================
0/0/0         g711alaw   y  S_CONNECT             FXSLS_CONNECT            
0/0/1         -          -  -                     FXSLS_ONHOOK             
0/0/2         -          -  -                     FXSLS_ONHOOK             
0/0/3         -          -  -                     FXSLS_ONHOOK             

3.      User 6101 can call user 6201 and the communication is normal.

Exception Handling

If the communication fails, run the debugging voice sipmsg all command to enable log collection of the sipmsg module and contact technical support personnel.

[Gateway Whiz] Interoperation cases (AR connected to devices such as Fortinet, Alcatel, Rais-ecom, and Cisco)