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 Phone Registers to CUCME/CUCM using SIP

IP Network

SIP Carrier

SIP Trunk from Carrier (ITSP)

Inter-Cluster SIP Trunk between two CUCM Clusters

IP Network
Inter-Cluster SIP Trunk between Gateway and CUCM/CUCME

IP Network

SIP Call Setup

SIP Gateway

SIP Gateway

IP
Calling Party

Invite (SDP)
100 Trying

180 Ringing

Called Party

SIP Signaling and SDP

(UDP or TCP)

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)

200 OK
ACK

RTP Stream
BYE

200 OK

Bearer or Media
(UDP)
Signaling

SIP Call Setup Using a Proxy Server

SIP Gateway

Proxy Server

SIP Gateway

IP
Calling Party
SIP Signaling and SDP
(UDP or TCP)

Invite
(SDP)
100
Trying
180
Ringing
200 OK
ACK

Invite (SDP)
100 Trying
180 Ringing
200 OK
ACK

RTP Stream

Bearer or Media
(UDP)
BYE

BYE

200 OK

200 OK

Called Party

Call Setup Using a Redirect Server

Redirect Server

SIP Gateway

SIP Gateway

IP
Invite
Calling Party

Called Party
Moved

SIP Signaling and SDP

(UDP or TCP)

Invite
Trying
Ringing
OK
ACK

Bearer or Media
(UDP)

RTP Stream
BYE
200 OK

SIP Addressing

Fully qualified domain names

sip:arasheed@abadnet.com.sa
E.164 addresses
sip:0114918199@gateway.com; user=phone
Mixed addresses
sip:0114918199; password=changeme@10.10.10.1 sip:arasheed@10.10.10.1

SIP Delayed Offer

IP
Calling Party

Invite
100 Trying
180 Ringing

Called Party
SIP Signaling and SDP
(UDP or TCP)

200 OK (SDP: Media Offer)

ACK (SDP: Media Answer)

RTP Stream
BYE
200 OK

Bearer or Media
(UDP)
Signaling

SIP Early Offer

IP
Calling Party

Invite (SDP: Media Offer)

100 Trying
180 Ringing

Called Party
SIP Signaling and SDP
(UDP or TCP)

200 OK (SDP: Media Answer)

ACK

RTP Stream
BYE
200 OK

Bearer or Media
(UDP)
Signaling

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 0
Router(config-if)# ip address 1.1.1.1 255.255.255.255
Router(config)# voice service voip
Router(conf-voi-serv)# sip
Router(conf-serv-sip)# session transport udp
Router(conf-serv-sip)# bind control source-interface Loopback 0
Router(conf-serv-sip)# bind media source-interface Loopback 0

Configuring SIP Gateways: SIP User Agent

sip.abadnet.com.sa

SIP ITSP
SIP Gateway

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

Configuring SIP Gateways

sip.abadnet.com.sa

Cisco Unified
Communications
Manager:
172.16.10.245

200.2.2.2

SIP ITSP
SIP Gateway

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 status

Displays 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 events

Verifies that the SIP subscription server is

up.

debug voip ccapi inout

Shows every interaction with the call control

API.

debug ccsip

For general SIP debugging; for example

views direction-attribute settings and port
and network address-translation traces.