SIP – A Technology Deep Dive

Anshu PrasadProduct Line Manager, MitelJune 2010

Laith ZalzalahDirector, Mitel NetSolutions

What is SIP?

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

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

ServiceProviders

Applications

slide 4

SIP Versus Other VoIP Protocols

slide 4

Open / Flexible Feature Rich Interoperable

SIP

H.323

MGCP /Megaco

Proprietary

slide 5

Components of a SIP System

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

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

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.

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 youfrom Jane!”

Jim is over at shopmart.com.

I’ll proxy the call over there!

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 youfrom Jane!”

sip.newplace.com“Try him at newplace.com”

“Call here for Jim from Jane”

slide 9

SIP Protocol

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

BYEBYE

200 OK 200 OK

Signaling Signaling

Media

slide 10

SIP Standards (RFCs)

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.

RFC 1321 The MD-5 Message Digest AlgorithmRFC 2617 HTTP Authentication : Basic and Digest AuthenticationRFC 2782 A DNS RR for specifying the location of services (DNS SRV)RFC 2976 The SIP INFO Method RFC 3262 Reliability of Provisional Responses in SIPRFC 3263 Locating SIP ServersRFC 3264 An Offer/Answer Model with SDP RFC 3265 SIP-Specific Event NotificationRFC 3311 The Session Initiation Protocol UPDATE Method RFC 3325 Asserted Identity within Trusted NetworksRFC 3329 Security Mechanism AgreementRFC 3515 The Session Initiation Protocol (SIP) Refer MethodRFC 3550 RTP: A Transport Protocol for Real-Time Applications

RFC 3551 RTP Profile for Audio and Video Conferences withMinimal ControlRFC 3665 Session Initiation Protocol Basic Call Flow ExamplesRFC 3725 Best Current Practices for 3rd Party Call ControlRFC 3824 Using E.164 numbers with the Session Initiation Protocol (SIP)RFC 3842 A Message Summary and Message Waiting Indication Event Package for the Session Initiation Protocol (SIP)RFC 3891 The Session Initiation Protocol (SIP) 'Replaces' HeaderRFC 3892 The SIP Referred-By Mechanism RFC 3966 The Tel URI for Telephone Numbers RFC 4028 Session Timers in the Session Initiation RFC 4566 SDP: Session Description Protocol RFC 4733 RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals

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 20138.5M

trunks24.3Mtrunks

* Data from Infonetics Research andFrost & Sullivan

2009 2016$717M $3.9B

2009 20163.8Musers

46Musers

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 telephonysystems

– Vendor benefits by developing only a single implementation

– Customer benefits by breadth of choice

SIP Trunk enterprise VoIP worldwide growth forecast*

slide 12

SIP Trunking Deployment Example

LAN

SIP PhonesSIP-Aware

Firewall

Enterprise

SIPSignaling

RTP Media

SIP SessionBorder Controller

PSTN GW

SoftSwitch

Broadband Link

Service ProviderSIP RegistrarSIP B2BUA

PSTN

LAN

slide 13

The Promise versus the Reality of SIP

Standards-based Numerous SIP RFCs still evolving– Most are open to interpretation– Some contain ambiguous or optional provisions

Successful interoperability can’t be assumed

Promise Reality

just because its SIP– Testing is required in every case as implementations

continue to evolve

Application developers are adopting SIP as their only telephony integration interface

– Proprietary integrations still exist in many cases

Cost savings are significant– … as long as interop issues are avoided through

careful testing prior to deployment in a production environment

Application integration

Vendor interoperability

Reduces costs

slide 14

Addressing the Challenges of SIPMitel SIP Center of Excellence

Validation of SIP interoperation with 3rd 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 3rd party test plans for 3rd party certification

– Detailed test results and configuration documentation produced for every interop

– Aligned and integrated with the Mitel Solutions Alliance partner ecosystem program

slide 15

Mitel SIP Qualified Interops

slide 16

Addressing the Challenges of SIPExample - Managed Services

slide 16

Current CLEC Certifications

Pending CLEC Certifications

Local Long Distance Wireless DataSIP Trunking

Fully integrated and optimized for Mitel’s CPE portfolio

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

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Successfully Deploying SIPTop 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 regularly4. 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 deployment10.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

slide 18

Conclusion

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

Mitel is fully committed to SIP– All Mitel PBX platforms offer SIP support and are extensively tested

with 3rd 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

ServiceProviders

Applications

Thank you

Anshu PrasadMitel Product Line Manageranshu_prasad@mitel.com(613) 592-2122 ext 2263