PRESENTACIÓN

 _{Signaling proxy}

The AR can serve as the IP PBX and process all registration and call messages of users. When receiving signaling messages from external users, the AR processes signaling messages and forwards them to the IMS (SBC). The AR is a user for the IMS (SBC). The IMS sends a call request to the AR. The AR processes the request, and then forwards it to the real called party.

 Media proxy

All media streams that are exchanged between users in on the internal network and external networks are processed and forwarded using the RTP/RTCP over the AR. The AR checks packet validity, and determines media stream forwarding policies including packet filter policy, QoS policy, and address translation policy based on the signaling process results. Then the AR specifies the address and port for receiving RTP streams of intranet and extranet users. In this way, media streams are forwarded properly and guaranteed good QoS and security in any networking.

Figure 10-2Processes in the SBC solution

CPE SBC

192.168.0.1 192.168.0.2 65.65.65.1 65.65.65.98 Media Proxy functon

enabled(SBC embeded) 192.168.0.1:10248 65.65.65.1:61448 INVITE Destination Addr:65.65.65.1 5060 SDP:65.65.65.98 42812 INVITE Destination Addr:192.168.0.1 5060 SDP:192.168.0.2 61440 200 OK Source Addr:192.168.0.1 5060 SDP:192.168.0.1 10248 200 OK Source Addr:65.65.65.1 5060 SDP:65.65.65.1 61448 RTP Source Addr:192.168.0.1 10248 Destination Addr:192.168.0.2 61440 RTP Source Addr:65.65.65.1 61448 Destination Addr:65.65.65.98 42812 RTP Source Addr:65.65.65.98 42812 Destin Addr:65.65.65.1 61448 RTP Source Addr:192.168.0.2 61440 Destin Addr:192.168.0.1 10248 65.65.65.98:42812 192.168.0.2:61440

1. The SBC sends an Invite message to the CPE. Assume that SDP carries the local media address 65.65.65.98, and the UDP port number is 42812. After receiving the SIP message, the SIP server/proxy on the CPE changes the SDP media address to a private network address 192.168.0.2, and changes the UDP port number to 61440. The SIP server/proxy then sends the Invite message to a SIP phone and records proxy mapping entries.

2. The SIP phone receives the Invite message, records the media address and port number 192.168.0.2:61440 of the peer end, and responds with a 200 OK message. In this case, the SDP carries the local media address 192.168.0.1, and the UDP port number is changed to 10248.

3. The CPE receives the 200 OK message from the SIP phone, changes the media address to the public network address 65.65.65.1, and changes the UDP port number to 61448. In addition, the CPE records the mapping the original media address and the new media address.

4. The message is sent from the SIP phone to the destination media address

192.168.0.2:61440. The CPE changes the destination media address to 65.65.65.98 42812 and sends the message to the SBC.

5. The SBC sends the message to the destination media address 65.65.65.1:61448. The CPE changes the phone destination media address to 192.168.0.1:10248 and sends the

Huawei AR G3 Series Enterprise Routers

Voice Feature White Paper 11 AR Voice Solution

11

AR Voice Solution

An AR G3 software package integrates all functions of the SIP AG and IP PBX, and can meet various networking requirements, such as voice gateway, call management, and trunk

interconnection. Features can be controlled using licenses. Different service licenses are selected to meet varying networking requirements. The following table lists license functions.

License Name Depends on Description

AR Voice

Value-added Service License

None This license is mandatory for all voice functions. Other licenses can be loaded only after this license is loaded.

 _{Provide the function of direction access by end}

users, including POTS and ISDN terminals.

 _{Provide the SIP AG function that uses the SIP}

protocol to process direct access from POTS and ISDN terminals.

 _{Support the TDM PBX VE1 access mode, and use}

the SIP protocol to process access from VE1 phones.

License Name Depends on Description CM&BEST License AR Voice Value-added Service License

The call manager (CM) feature implements user management and call management, for POTS phones, ISDN phones, and SIP UEs. This feature supports all new services of the PBX. Some new services, such as IVR, require service licenses. Trunks, such as AT0, E1, and SIP, are supported.

The BEST feature provides internal call and inter-office call processing of the local branch when the IP WAN is faulty. The BEST feature supports only basic call functions, and does not support PBX value-added services.

 _{In a branch, the BEST feature can provide call}

service for analog phones, IP phones or TDM PBXs.

 _{Inter-office calls can be routed through a SIP trunk}

to the IP WAN or through an AT0 or E1 trunk to the PSTN.

 _{The E1/AT0 trunk takes effect only when the SIP}

trunk is faulty. When the SIP trunk recovers, inter-office calls are no longer routed through the E1/AT0 trunk.

 _{When the SIP trunk is working normally, calls in a}

branch are processed by the headquarters CM after being routed to the headquarters CM through the SIP trunk. CT (Call Trunk) License AR Voice Value-added Service License

The call trunk (TG) feature provides interconnection between trunks and converts packets of different protocols. It does not provide direct access to terminals.

 _{Support AT0, E1, and SIP trunk ports. These trunk}

ports are peer-to-peer for the CT.

 _{Support the conversion from H.323 to SIP and from}

R2 to SIP and the codec conversion. IVR (Interactive Voice Response) License CM&BEST License or CT (Call Trunk) License

The IVR automatic connection service is similar to an attendant position. When a user dials the IVR service access code, the IVR system plays a voice message, promoting the user to dial an extension number or the main number. The user can dial the extension number to start a basic call, or dial the main number to trigger the simultaneous ringing, sequential ringing, line selection (selecting the next user in accordance with certain rules) services. When multiple users call the IVR service, subsequent users wait in the call queue and hear the call queuing announcement.

Huawei AR G3 Series Enterprise Routers

Voice Feature White Paper 11 AR Voice Solution