SIP – A Technology Deep Dive
Anshu Prasad
Product Line Manager, Mitel
June 2010
Laith Zalzalah
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
slide 4
SIP Versus Other VoIP Protocols
slide 4
Open / Flexible
Feature Rich
Interoperable
SIP
H.323
MGCP /
Megaco
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
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 you
from 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 you
from Jane!”
sip.newplace.com
“Try him at
newplace.com”
“Call here
for Jim
from Jane”
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
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
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
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 LinkService Provider
SIP Registrar SIP B2BUA PSTN LANslide 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
slide 14
Addressing the Challenges of SIP
Mitel SIP Center of Excellence
Æ
Validation of SIP interoperation with 3
rdparty
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
rdparty test plans for 3
rdparty
certification
–
Detailed test results and configuration
documentation produced for every interop
–
Aligned and integrated with the Mitel Solutions
Alliance partner ecosystem program
slide 15
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
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 tomap your requirements against the features supported
5.
Review the interop documentation
– Confirm known limitations won’t affectyour 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 workingprior 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
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