Signaling Protocol: SIP

Introduction to SIP Protocol

Defined in IETF RFCs.

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.

Simple extensible protocol.

Supports audio, video, and data.

Introduction to SIP Protocol (Cont..)

The main signaling functions of SIP are as follows:

Location of an end point;

Contacting an end point to determine willingness to establish a session;

Exchange of media information to allow a session to be established;

Modification of existing media sessions;

Teardown of existing media sessions.

Benefits of SIP Protocol

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 Components

User Agents: Peers in a session. Two types: User Agent Server and User Agent Client.

User Agent Client: Client application that initiate a request.

User Agent Server: Server application that contact the user when an INVITE message is received and

then Send the response.

SIP components can be classified as Clients and Servers

Clients (Endpoints)

Phones: An IP telephone work a UAC or UAS.

Gateway: Works as a UAS or UAC and provides call control support. Performs translation between PSTN

and VoIP networks.

Servers: Registrar, proxy, redirect, and location.

SIP Servers

Registration Server: Receives requests from UACs for registration of their current location.

Proxy server: An intermediate component that receives SIP requests from a client

and then forwards the requests on behalf of the client to the next SIP server in the network.

Redirect Server: Provides the client with information of the next hop or hops that the message should take.

Location Server: Implement mechanisms to resolve addresses.

SIP in Cisco Unified Communications

IP Network

IP Network

SIP Carrier

IP Network

IP Phone Registers to CUCME/CUCM using SIP

SIP Trunk from Carrier (ITSP)

Inter-Cluster SIP Trunk between two CUCM Clusters

Inter-Cluster SIP Trunk between Gateway and CUCM/CUCME

SIP Call Setup

SIP 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

SDP (Session Description Protocol):

Used to exchange media capabilities.

Sent with INVITE message (early-offer)

or 200 OK message by the called party

(delayed-offer)

SIP Call Setup Using a Proxy Server

Proxy ServerSIP Gateway

IP

SIP Signaling and SDP

(UDP or TCP)

Bearer or Media

(UDP)

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 Server

Redirect ServerSIP Gateway

IP

SIP Signaling and SDP

(UDP or TCP)

Bearer or Media

(UDP)

RTP Stream

Ringing

OK

Invite

Trying

ACK

BYE

200 OK

SIP Gateway

Calling Party Called PartyInvite

Moved

SIP Addressing

Fully qualified domain namessip:arasheed@abadnet.com.sa

E.164 addressessip:0114918199@gateway.com; user=phone

Mixed addressessip:0114918199; password=changeme@10.10.10.1 sip:arasheed@10.10.10.1

SIP Delayed Offer

IP

SIP Signaling and SDP

(UDP or TCP)

Bearer or Media

(UDP)

RTP Stream

Signaling

180 Ringing

200 OK (SDP: Media Offer)

Invite

100 Trying

ACK (SDP: Media Answer)

BYE

200 OK

Calling Party Called Party

SIP Early Offer

IP

SIP Signaling and SDP

(UDP or TCP)

Bearer or Media

(UDP)

RTP Stream

Signaling

180 Ringing

200 OK (SDP: Media Answer)

Invite (SDP: Media Offer)

100 Trying

ACK

BYE

200 OK

Calling Party Called Party

Configuring SIP Gateways

Enable SIP voice services

Configure SIP service

Transport

Bind interface

Configure SIP User Agent (UA)

Authentication

SIP servers

Configure dial-peer SIP parameters

Session protocol

Session target

Configuring SIP Gateways: Session Transport and Bind Interface

Router(config)#interface loopback 0Router(config-if)# ip address 1.1.1.1 255.255.255.255

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

Configuring SIP Gateways: SIP User Agent

Router(config)# sip-ua

Router(config-sip-ua)# authentication username arasheed password secret

Router(config-sip-ua)# registrar dns:sip.abadnet.com.sa expires 3600

Router(config-sip-ua)# sip-server dns:sip.abadnet.com.sa

SIP ITSP

SIP Gateway

sip.abadnet.com.sa

Configuring SIP Gateways

SIP ITSP

SIP Gateway

sip.abadnet.com.saCisco Unified

Communications

Manager:

172.16.10.245

200.2.2.2

Router(config)# dial-peer voice 2000 voip

Router(config-dial-peer)# destination-pattern 1...

Router(config-dial-peer)# session protocol sipv2

Router(config-dial-peer)# session target sip-server

Router(config)# dial-peer voice 2001 voip

Router(config-dial-peer)# destination-pattern 1...

Router(config-dial-peer)# session protocol sipv2

Router(config-dial-peer)# session target ipv4:172.16.10.245

Router(config-dial-peer)# preference 1

Router(config)# dial-peer voice 90 voip

Router(config-dial-peer)# destination-pattern 9T

Router(config-dial-peer)# session target ipv4:200.2.2.2

Router(config-dial-peer)# session protocol sipv2

Verifying SIP Gateway Configuration

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.

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