© 2006 Cisco Systems, Inc. All rights reserved.GWGK v2.01-1 Gateway Deployments Implementing SIP Gateways. - презентация

1 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Gateway Deployments Implementing SIP Gateways

2 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Overview of SIP Gateways SIP was developed by the IETF for multimedia conferencing over IP. SIP provides nonproprietary advantages in these areas: –Protocol extensibility –System scalability –Personal mobility services –Interoperability with different vendors

3 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Overview of SIP Gateways (Cont.) SIP provides these capabilities: Determines the location of the target endpoint Determines the media capabilities of the target endpoint Determines the availability of the target endpoint Establishes a session between the originating and target endpoints Handles the transfer and termination of calls

4 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Call Flow SIP Gateway PSTN Signaling Bearer or Media RTP Stream Signaling Invite Trying Ringing OK Invite Trying Ringing OK ACK BYE OK

5 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Why SIP? Advantages of SIP gateways: Dial-plan configuration directly on the gateway Translations defined per gateway Advanced support for third-party telephony system integration Interoperability with third-party voice gateways Support of third-party end devices (SIP phones)

6 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Why SIP? (Cont.) H.323 versus MGCP versus SIP gateways: H.323MGCPSIP Pros Dial plan directly on the gateway Translations defined per gateway Regional require- ments can be met More specific call routing Advanced fax support Centralized dial plan configuration Centralized gateway configuration Simple gateway configuration Easy implementation Support of QSIG supplementary services Dial plan directly on the gateway Translations defined per gateway Third-party telephony system support Third-party gateway interoperability Third-party end-device support ConsComplex configuration Extra SRST-related call routing configuration Less feature support SIP support depends on CCM version

7 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Integration Options Cisco Unified CallManager SIP Proxy Cisco Unity Express Cisco SIP IP Phone Provider SIP Network SIP Voice Gateway/ Cisco Unified CallManager Express

8 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Integration Options (Cont.) SIP integration with Cisco IOS: Cisco Unified CallManager support –Starting with version 4.0, Cisco Unified CallManager supports SIP gateways and SIP trunks. SIP proxy server support –Cisco IOS supports SIP, so it can be used as gateway by Cisco SIP proxy server and third-party SIP servers. SIP VoIP support –Cisco IOS supports SIP, so it can be connected to a SIP VoIP service provider network.

9 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Integration Options (Cont.) SIP integration with Cisco IOS: SIP SRST support –SRST Version 3.4 on Cisco IOS supports SRST for SIP phones on Cisco Unified CallManager 5.0. Cisco Unity Express support –Cisco Unity Express supports only SIP as the signaling protocol.

10 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Configuration Considerations SIP DTMF on Cisco Unified CallManager: SIP DTMF requires MTP on Cisco Unified CallManager. SCCP Phone SIP Gateway MTP Out-of-Band SCCP Out-of-Band Information In-Band SIP RTP

11 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v SIP Configuration Considerations (Cont.) SIP DTMF on IOS gateways: SIP DTMF relay is configured on gateways in dial peer configuration mode. There are two methods: –RTP NTE: Forwards DTMF tones by using RTP with the NTE payload type –SIP NOTIFY: Forwards DTMF tones using SIP NOTIFY messages SCCP IP phones only support out-of band. Therefore SIP NOTIFY must be used.

12 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v voice service voip Router(config)# Enters voice service VoIP configuration mode sip router(conf-voi-serv)# Enables SIP signaling for the VoIP service and enters SIP configuration mode SIP Commands Enabling voice service on the gateway:

13 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v session transport {tcp|udp} Router(conf-serv-sip)# Defines the transport protocol for SIP bind {all|control|media} source-interface interface router(conf-serv-sip)# Defines the source interface for SIP packets SIP Commands (Cont.) Configuring SIP service parameters:

14 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v sip-ua Router(config)# Enters SIP UA configuration authentication username username password password router(config-sip-ua)# Defines the username and password used to connect to the SIP server SIP Commands (Cont.) Configuring SIP UA parameters: registrar {dns:server|ip:ip-address} expires seconds router(config-sip-ua)# Configures the name or IP address of the SIP server

15 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v session protocol sipv2 Router(config-dial-peer)# Specifies SIP to be the protocol used on the VoIP dial peer dtmf-relay rtp-nte router(config-dial-peer)# Configures DTMF to be transferred in-band into RTP SIP Commands (Cont.) Configuring SIP dial-peer parameters: dtmf-relay sip-notify router(config-dial-peer)# Configures DTMF to be transferred out-of-band via sip-notify

16 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Integrating IOS Gateways with SIP VoIP Networks Perform at least these steps to connect Cisco Unified CallManager Express to a provider SIP network: 1. Enable SIP and specify SIP parameters. 2. Configure SIP UA and VoIP dial peer to the provider.

17 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Integrating IOS Gateways with SIP VoIP Networks: Scenario PSTN Cisco Unified CallManager Express Configure Cisco Unified CallManager Express to connect to a SIP service provider network and route external calls via that connection.

18 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Integrating IOS Gateways with SIP VoIP Networks: Configure the SIP Service PSTN Cisco Unified CallManager Express voice service voip sip session transport udp bind control source-interface FastEthernet0/0 bind media source-interface FastEthernet0/0 no shutdown ! interface FastEthernet0/0 ip address no shutdown

19 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Integrating IOS Gateways with SIP VoIP Networks: Configure the VoIP Connection PSTN Cisco Unified CallManager Express sip-ua authentication username JohnDoe password A2E2421 registrar dns:sip.cisco.com expires 3600 ! dial-peer voice 2 voip destination-pattern 9T session protocol sipv2 session target dns:sip.cisco.com incoming called-number 2... codec g711alaw

20 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v show sip-ua calls router# Displays active SIP UA calls show sip-ua connections {tcp|udp} {brief|detail} router# Displays active SIP UA connections Verifying an SIP Integration SIP UA verification commands: show sip-ua statistics router# Displays SIP UA statistics

21 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v show dial-peer voice tag router# Displays detailed information about the specified dial peer show dial-peer voice summary router# Displays a summary of all active dial peers Verifying a SIP Integration (Cont.) Dial peer show and debug commands: debug voip dialpeer router# Displays default debug output for all active VoIP dial peers

22 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Verifying a SIP Integration (Cont.) SIP show commands: show sip service router# Displays the status of the SIP VoIP service show sip-ua statistics router# Displays SIP traffic statistics show sip-ua status router# Displays the status of the SIP UA

23 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Summary SIP is an IETF standard that is ideal for IOS voice gateways within third-party VoIP networks. A SIP call flow consists of signaling and transmission of bearer and media packets. SIP, like H.323, allows extremely flexible decentralized call processing and call handling. SIP gateways and Cisco Unified CallManager Express can be integrated in Cisco Unified CallManager, SIP Proxy, Cisco Unity Express, SRST, and SIP voice provider networks. SIP needs an MTP to support DTMF on Cisco Unified CallManager.

24 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v Summary (Cont.) There are several commands available on Cisco IOS to configure SIP on Cisco IOS routers. To integrate Cisco Unified CallManager Express into SIP VoIP service provider networks, SIP service, SIP UA, and SIP dial peers must be configured. There are several commands available on Cisco IOS to verify a SIP integration.

25 © 2006 Cisco Systems, Inc. All rights reserved.GWGK v