Implementing SIP Gateways

© 2008 Cisco Systems, Inc. All rights reserved. CVOICE v6.0—3-1

Implementing VoIP Gateways

Implementing SIP Gateways


SIP Fundamentals

SIP is a simple extensible protocol.

SIP is defined in IETF RFC 2543 and RFC 3261.

SIP creates, modifies, and terminates multimedia sessions with one or more participants.

SIP leverages various standards: RTP, RTCP, HTTP, SDP, DNS, SAP, MGCP, and RTSP.

SIP performs addressing by E.164, e-mail, or DNS service record.

SIP is ASCII text-based for easy implementation and debugging.


SIP Fundamentals (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


How SIP Works

ASCI-based protocol

User identified by a unique SIP address

– sip:[email protected]

Users register with registrar server using their assigned SIP addresses

When user initiates a call, a SIP request is sent to a SIP server

Location of end user can be dynamically registered with the SIP server


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)


SIP Architecture

SIP Gateway

Legacy PBX

PSTN

SIP

RTP

SIPSIP

SIP Proxy, Registrar,

Location, and Redirect Servers

SIP User Agents (UAs)

T1 or PRI


SIP Call FlowSIP Gateway

PSTN

Signaling

Bearer or MediaRTP Stream

Signaling

Invite (SDP)

100 Trying

180 Ringing

200 OK

Invite (SDP)

100 Trying

180 Ringing

200 OK

ACK

BYE

200 OK

SIP Proxy Server

ACK


Direct Call SetupSIP Gateway

IP

SIP Signaling and SDP(UDP or TCP)

Bearer or Media (UDP)

RTP Stream

Signaling

180 Ringing

200 OK

Invite (SDP)

100 Trying

ACK

BYE

200 OK

SIP Gateway

Calling Party Called Party


Call Setup Using a Proxy ServerProxy Server

SIP Gateway

IP



RTP Stream

180 Ringing

200 OK

Invite (SDP)

100 Trying

ACK

BYE

200 OK

SIP Gateway

Calling Party Called PartyInvite (SDP)

100 Trying

180 Ringing

200 OK

ACK

BYE

200 OK


Call Setup Using a Redirect ServerRedirect Server

SIP Gateway

IP



RTP Stream

Ringing

OK

Invite

Trying

ACK

BYE

200 OK

SIP Gateway

Calling Party Called PartyInvite

Moved


SIP Addresses

Fully qualified domain names

– sip:[email protected]

E.164 addresses

– sip:[email protected]; user=phone

Mixed addresses

– sip:14085551234; [email protected] sip:[email protected]


Address Registration

SIP Gateway

Location Database

Registrar Server

Redirect Server

SIP UACs

SIP Proxy (UAS)

SIP UACs

RegisterHere I am!


Address Resolution

Location Database

Registrar Server

Redirect Server

SIP Proxy

Where is the name or phone number?


SIP DTMF Configuration Considerations

SCCP Phone

SIP Gateway

MTP

Out-of-Band SCCP

Out-of-Band Information

In-Band SIP

RTP RTP

SIP DTMF on Cisco Unified Communications Manager SIP DTMF requires MTP on Cisco Unified Communications

Manager.


SIP DTMF Considerations (Cont.)

SIP DTMF on Cisco IOS gateways: SIP DTMF relay is configured on gateways in dial-peer

configuration mode. There are two methods:

– RTP Named Telephony Event: 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.


Configuring a SIP Gateway

Enable SIP voice services

Configure SIP service

– Transport

– Bind interface

Configure SIP User Agent (UA)

– Timers

– Authentication

– SIP servers

Configure dial-peer SIP parameters

– Session protocol

– Session target

– DTMF relay


SIP ITSP

Configure Cisco Unified Communications Manager Express to connect to a SIP service provider

network and route external calls via that connection.

router(config)# voice service voiprouter(conf-voi-serv)# siprouter(conf-serv-sip)# session transport udprouter(conf-serv-sip)# bind control source-interface Loopback 0router(conf-serv-sip)# bind media source-interface Loopback 0

SIP Gateway

Sip2.cisco.com

Integrating Cisco IOS Gateways with a SIP ITSP


router(config)# sip-uarouter(config-sip-ua)# authentication username JDoe password secretrouter(config-sip-ua)# registrar dns:sip2.cisco.com expires 3600router(config-sip-ua)# sip-server dns:sip2.cisco.comrouter(config-sip-ua)# retry invite 2router(config-sip-ua)# retry response 2router(config-sip-ua)# retry bye 2router(config-sip-ua)# retry cancel 2

SIP ITSP

SIP Gateway

Sip2.cisco.com

Integrating Cisco IOS Gateways with a SIP ITSP (Cont.)


SIP ITSP

Sip2.cisco.com

Ext.: 2…

router(config)# dial-peer voice 2000 voiprouter(config-dial-peer)# destination-pattern 2...router(config-dial-peer)# session protocol sipv2router(config-dial-peer)# session target sip-serverrouter(config-dial-peer)# dtmf-relay rtp-nte router(config)# dial-peer voice 2001 voiprouter(config-dial-peer)# destination-pattern 2...router(config-dial-peer)# session protocol sipv2router(config-dial-peer)# session target ipv4:10.1.1.15router(config-dial-peer)# dtmf-relay sip-notifyrouter(config-dial-peer)# preference 1router(config)# dial-peer voice 90 voiprouter(config-dial-peer)# destination-pattern 9Trouter(config-dial-peer)# session target ipv4:192.168.1.100router(config-dial-peer)# session protocol sipv2 router(config-dial-peer)# dtmf-relay rtp-nte

Cisco Unified Communications

Manager: 10.1.1.15

SIP Gateway

192.168.1.100

Integrating Cisco IOS Gateways with a SIP ITSP (Cont.)


Command Description

show sip-ua service Displays the status of the SIP VoIP service.

show sip-ua status Displays the status of the SIP UA.

show sip-ua register statusDisplays the status of E.164 numbers that a SIP gateway has registered with an external primary SIP registrar.

show sip-ua timers Displays SIP UA timers.

show sip-ua connections Displays active SIP UA connections.

show sip-ua calls Displays active SIP UA calls.

show sip-ua statistics Displays SIP traffic statistics.

Verifying SIP Gateways


Router# show sip serviceSIP Service is up

Router# show sip-ua statusSIP User Agent StatusSIP User Agent for UDP : ENABLEDSIP User Agent for TCP : ENABLEDSIP User Agent bind status(signaling): DISABLEDSIP User Agent bind status(media): DISABLEDSIP max-forwards : 6SIP DNS SRV version: 1 (rfc 2052)Redirection (3xx) message handling: ENABLED

Router# show sip-ua timersSIP UA Timer Values (millisecs)trying 500, expires 180000, connect 500, disconnect 500comet 500, prack 500, rel1xx 500, notify 500refer 500, register 500

Verifying SIP Gateways (Cont.)


Verifying SIP Gateways (Cont.)

Router# show sip-ua register status

Line peer expires(sec) registered

4001 20001 596 no

4002 20002 596 no

5100 1 596 no

9998 2 596 no


Verifying SIP Gateways (Cont.)router# show sip-ua callsSIP UAC CALL INFO Number of SIP User Agent Client(UAC) calls: 0

SIP UAS CALL INFOCall 1SIP Call ID : [email protected] State of the call : STATE_ACTIVE (7) Substate of the call : SUBSTATE_NONE (0) Calling Number : 2818902001 Called Number : 1003 Bit Flags : 0x1212003A 0x100000 0x488 CC Call ID : 1 Source IP Address (Sig ): 10.10.10.1 Destn SIP Req Addr:Port : 10.10.10.2:5060 Destn SIP Resp Addr:Port: 10.10.10.2:56884 Destination Name : 10.10.10.2..


Verifying SIP Gateways (Cont.).. Number of Media Streams : 1 Number of Active Streams: 1 RTP Fork Object : 0x0 Media Stream 1 State of the stream : STREAM_ACTIVE Stream Call ID : 1 Stream Type : voice-only (0) Negotiated Codec : g729r8 (20 bytes) Codec Payload Type : 18 Negotiated Dtmf-relay : inband-voice Dtmf-relay Payload Type : 0 Media Source IP Addr:Port: 10.10.10.1:18050 Media Dest IP Addr:Port : 10.10.10.2:16522 Orig Media Dest IP Addr:Port : 0.0.0.0:0

Number of SIP User Agent Server(UAS) calls: 1


Debug Commands

Command Description

debug asnl eventsVerifies that the SIP subscription server is up.

debug voip ccapi inoutShows every interaction with the call control API.

debug voip ccapi protoheadersDisplays messages sent between the originating and terminating gateways.

debug ccsip

For general SIP debugging; for example views direction-attribute settings and port and network address-translation traces.


HQ-1# debug ccsip messages

SIP Call messages tracing is enabled HQ-1# *Mar 6 14:19:14: Sent: INVITE sip:[email protected];user=phone;phone-context=unknownSIP/2.0 Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:[email protected]> To: <sip:[email protected];user=phone;phone-context=unknown> Date: Sat, 06 Mar 1993 19:19:14 GMT Call-ID: [email protected] Cisco-Guid: 2881152943-2184249568-0-483551624 User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled CSeq: 101 INVITE Max-Forwards: 6 Timestamp: 731427554Contact: <sip:[email protected]:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 138

Debug Commands (Cont.)


*Mar 6 14:19:16: Received: SIP/2.0 200 OK Via: SIP/2.0/UDP 166.34.245.230:55820 From: "3660110" <sip:[email protected]> To: <sip:[email protected];user=phone;phone-context=unknown>;tag=27DBC6D8-1357 Date: Mon, 08 Mar 1993 22:45:12 GMT Call-ID: [email protected] Timestamp: 731427554 Server: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled Contact: <sip:[email protected]:5060;user=phone> CSeq: 101 INVITE Content-Type: application/sdp Content-Length: 138

v=0 o=CiscoSystemsSIP-GW-UserAgent 1193 7927 IN IP4 166.34.245.231s=SIP Call t=0 0 c=IN IP4 166.34.245.231 m=audio 20224 RTP/AVP 0




*Mar 6 14:19:19: Received: BYE sip:[email protected]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 166.34.245.231:53600 From: <sip:[email protected];user=phone;phone-context=unknown>;tag=27DBC6D8-1357 To: "3660110" <sip:[email protected]> Date: Mon, 08 Mar 1993 22:45:14 GMT Call-ID: [email protected] User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabledMax-Forwards: 6 Timestamp: 731612717 CSeq: 101 BYE Content-Length: 0


Summary

SIP is defined by IETF RFC 2543 and RFC 3261 and allows easy integration with third-party VoIP networks.

There are five advantages to using SIP gateways as voice gateways.

SIP is modeled on the interworking of UAs and network servers.

A SIP call flow consists of signaling and transmitting bearer and media packets.


Summary (Cont.)

A SIP address consists of an optional user ID, a host description, and optional parameters to qualify the address more precisely.

SIP call-setup models include direct, proxy server, and redirection.

There are seven commands available on Cisco IOS to configure SIP on Cisco IOS routers.

There are at least thirteen commands available on Cisco IOS to verify and troubleshoot a SIP integration.


VoiceGateway

Voice-Enabled Router

IP Telephony Router with Cisco Unified Communications Manager Express

Cisco Unified Communications Manager

Switch

Router

POTS Phone

IP Phone

Network Cloud

Line: Serial Line: Ethernet

PC

PBX

Cisco Unified Border Element

IP

Icons

Implementing SIP Gateways

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