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SIP – A Technology Deep Dive

Anshu Prasad

Product Line Manager, Mitel

June 2010

Laith Zalzalah

(3)

What is SIP?

Æ

Standardized by the Internet Engineering Task Force (IETF) in

Request for Comment (RFC) documents

The model is lightweight – the standard focuses on initiating, modifying

and terminating sessions

The model is decentralized – its up to the parties establishing the

session to negotiate the attributes of the session

slide 3

The power of SIP is that it is Flexible, Extensible and Open

Æ

Session Initiation Protocol (SIP) is a

signaling protocol for establishing

sessions

We most commonly think of Voice over IP

(VoIP), but applies to multimedia/video,

instant messaging, faxing, web

integration and more

SIP PBX SIP Devices Service Providers Applications

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slide 4

SIP Versus Other VoIP Protocols

slide 4

Open / Flexible

Feature Rich

Interoperable

SIP

H.323

MGCP /

Megaco

(5)

slide 5

Components of a SIP System

Æ

Logically, there are several parts to a SIP system

User Agent (UA) – the phone or server

Registrar – keeps track of where the user is within a system

Redirect Server – used to inform devices when they need to contact

different locations

Proxy Server – used to relay messages back and forth within the system

Æ

In practice, several functions may actually be in the same server

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slide 6

SIP Walkthrough – Registration

SIP Server

sip.telco.com

“Lunch time! Use my

mobile at 10.0.1.1”

“I’m Jane, and I’ll be

using a phone at

192.168.0.1 today!”

192.168.0.1

10.0.1.1

When I get a call

for Jane, I’ll know

to contact her at

192.168.0.1!

Ah, now I will have

to use 10.0.1.1 to

reach Jane.

(7)

slide 7

SIP Walkthrough – Proxy

slide 7

sip.telco.com

sip.shopmart.com

“I need to call Jim!”

“Call here for

Jim from Jane”

“Call for you

from Jane!”

Jim is over at

shopmart.com.

I’ll proxy the

call over there!

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slide 8

SIP Walkthrough – Redirection

slide 8

sip.telco.com

sip.shopmart.com

“I need to

call Jim!”

“Call here for

Jim from Jane

“Call for you

from Jane!”

sip.newplace.com

“Try him at

newplace.com”

“Call here

for Jim

from Jane”

(9)

slide 9

SIP Protocol

Æ

SIP is much like the HTTP request/response model

Uses a text-based message containing a header and message body

Invite Invite 100 Trying 180 Ringing 200 OK ACK 180 Ringing 200 OK ACK RTP Media BYE BYE 200 OK 200 OK

Signaling

Signaling

Media

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slide 10

SIP Standards (RFCs)

Æ

RFC 3261 SIP: Session Initiation Protocol

SIP utilizes other protocols: TCP or UDP for connection, Real-time

Transport Protocol (RTP) to carry media, Session Description Protocol

(SDP) to define parameters of the RTP, etc.

Æ RFC 1321 The MD-5 Message Digest Algorithm

Æ RFC 2617 HTTP Authentication : Basic and Digest Authentication Æ RFC 2782 A DNS RR for specifying the location of services (DNS

SRV)

Æ RFC 2976 The SIP INFO Method

Æ RFC 3262 Reliability of Provisional Responses in SIP Æ RFC 3263 Locating SIP Servers

Æ RFC 3264 An Offer/Answer Model with SDP Æ RFC 3265 SIP-Specific Event Notification

Æ RFC 3311 The Session Initiation Protocol UPDATE Method Æ RFC 3325 Asserted Identity within Trusted Networks Æ RFC 3329 Security Mechanism Agreement

Æ RFC 3515 The Session Initiation Protocol (SIP) Refer Method Æ RFC 3550 RTP: A Transport Protocol for Real-Time Applications

Æ RFC 3551 RTP Profile for Audio and Video Conferences with Minimal Control

Æ RFC 3665 Session Initiation Protocol Basic Call Flow Examples Æ RFC 3725 Best Current Practices for 3rd Party Call Control

Æ RFC 3824 Using E.164 numbers with the Session Initiation Protocol (SIP)

Æ RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)

Æ RFC 3891 The Session Initiation Protocol (SIP) 'Replaces' Header Æ RFC 3892 The SIP Referred-By Mechanism

Æ RFC 3966 The Tel URI for Telephone Numbers Æ RFC 4028 Session Timers in the Session Initiation Æ RFC 4566 SDP: Session Description Protocol

Æ RFC 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

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slide 11

SIP Trunking, Stations and Applications

SIP trunking to a service

provider network

By being “virtual” trunks, cost is significantly reduced compared to legacy physical digital or analog trunks

Network simplification is achieved by combining voice and data network services on IP

2009

2013

8.5M

trunks

24.3M

trunks

* Data from Infonetics Research and Frost & Sullivan

2009

2016

$717M

$3.9B

2009

2016

3.8M

users

46M

users

SIP line/station support

Avoids vendor lock-in;

customer can pick and

choose the devices they want

Functionality specific

endpoints can be more easily integrated

conference phones, video phones, mobile or desktop softphones, etc.

SIP applications

No need for proprietary API’s or custom development to achieve business process integration with telephony systems

Vendor benefits by

developing only a single implementation

Customer benefits by breadth of choice

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slide 12

SIP Trunking Deployment Example

LAN SIP Phones SIP-Aware Firewall

Enterprise

SIP Signaling RTP Media SIP Session Border Controller PSTN GW SoftSwitch Broadband Link

Service Provider

SIP Registrar SIP B2BUA PSTN LAN

(13)

slide 13

The Promise versus the Reality of SIP

Æ

Standards-based

Æ

Numerous SIP RFCs still evolving

Most are open to interpretation

Some contain ambiguous or optional provisions

Æ

Successful interoperability can’t be assumed

Promise

Reality

just because its SIP

Testing is required in every case as implementations continue to evolve

Æ

Application developers are adopting SIP as their

only telephony integration interface

Proprietary integrations still exist in many cases

Æ

Cost savings are significant

… as long as interop issues are avoided through careful testing prior to deployment in a production environment

Æ

Application integration

Æ

Vendor interoperability

(14)

slide 14

Addressing the Challenges of SIP

Mitel SIP Center of Excellence

Æ

Validation of SIP interoperation with 3

rd

party

network devices, endpoints, applications and

service provider networks

Dedicated lab capable of executing on-site or

remote SIP interop tests

Standardized test plans for both trunk and line

testing

Test execution of 3

rd

party test plans for 3

rd

party

certification

Detailed test results and configuration

documentation produced for every interop

Aligned and integrated with the Mitel Solutions

Alliance partner ecosystem program

(15)

slide 15

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slide 16

Addressing the Challenges of SIP

Example - Managed Services

slide 16

Current CLEC Certifications

Pending CLEC Certifications

Local Long Distance Wireless Data

SIP Trunking

Æ

Fully integrated and optimized for Mitel’s CPE portfolio

Æ

Engineered to replicate PSTN feature set while providing the benefits

of IP-based SIP trunking

More efficiently allocate and utilize bandwidth

Centralize call control across geographic boundaries

Provides address-based E911 coverage for every registered site to their local

Public Safety Responders

Improves survivability with business continuity routing

Enables cost containment by centralizing network support and infrastructure

NetSolutions and the Mitel Communications Platforms –

“Single Point of Accountability” Managed Services Offering

(17)

slide 17

Successfully Deploying SIP

Top 10 Tips

1.

Ensure your network infrastructure is

capable of supporting VoIP

Voice quality issues, connectivity failure, etc. are often due to the underlying network

2.

Choose your solution provider

carefully

Need to be SIP knowledgeable in order to diagnose and debug issues

efficiently

3.

Confirm interop compatibility

Version matching is important as SIP implementations change regularly

4.

Understand your feature requirements

SIP capabilities vary, so its critical to

map your requirements against the features supported

5.

Review the interop documentation

Confirm known limitations won’t affect

your deployment

6.

For SIP trunking, ensure you have a

qualified SIP-aware firewall

Most SIP trunking issues start and end here

7.

For SIP devices, consider education

and training for your users

Keep in mind that SIP devices typically behave differently than legacy phones

8.

Evaluate scalability and manageability

of the solution

Deploying a single device is very different from deploying 500 or 1000

9.

Do a trial deployment prior to cutover

Test important functionality is working

prior to mass deployment

10.

Don’t assume too much

Interop testing is limited; just because two things interop doesn’t mean they’ll work the way you need them to

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slide 18

Conclusion

Æ

SIP is the present and future of IP-based

communications

Vendor implementations are maturing and

normalizing

SIP trunking is very common and customers

are reaping significant benefit today

SIP device and application integration

continues to improve rapidly

slide 18

Æ

Mitel is fully committed to SIP

All Mitel PBX platforms offer SIP support and are extensively tested

with 3

rd

party implementations

The Mitel Border Gateway product provides SIP-aware firewall

functionality in addition to other advanced services

Mitel NetSolutions offers fully managed communications services

including SIP

SIP PBX SIP Devices Service Providers Applications

(19)

Thank you

Anshu Prasad

Mitel Product Line Manager

anshu_prasad@mitel.com

(613) 592-2122 ext 2263

References

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