SIP – Yesterday, Today, & Tomorrow

Jon Murphy

Sr. Network Application Engineer

tw telecom

Introduction

Jon Murphy

Sr. Network Application Engineer

tw telecom

jon.murphy@twtelecom.com

(614) 255-2132 (office)

(614) 313-6925 (cell)

GOAL/Agenda

I hope you leave here today understanding:

1) What is SIP? Overall Concept, Definition, and Components.

2) How did SIP get here? History of SIP/VOIP

3) Why SIP/VOIP?

4) What does the future of SIP/VOIP look like?

A little history, a little overview, a little tech, a little bit of everything…

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Warning No “Commercials” • Lets build a “SIP” hamburger

with minimal “bun”!

• I will try to add some spice with pickles and tomatoes but at the end of the day this is still “SIP”… (or is it “SIP with SIZZLE” – more to come)…

• I will commit to you to try to stick to the “meat” of SIP without “the cheese” of course”

• Please feel free to make this an interactive as possible!

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Start with a “Knowledge Foundation”

• VOIP is a family of technologies, methodologies, communication protocols, and transmission techniques for the delivery of voice communications and multimedia sessions over IP networks, such as the Internet for example.

• Session Initiation Protocol (SIP) is an signaling protocol for VOIP for creating, modifying, and terminating sessions with one or more participants of a VOIP call. Other well know signaling protocols are MGCP, H.323, SKINNY for examples

• H.323 a call control element and signaling protocol that provides service to telephones or videophones. Such a device may provide or facilitate both basic services and supplementary services, such as call transfer, park, pick-up, and hold. IP-based PBX might be an H.323 Gatekeeper for example

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School’s still in: More Basic VOIP Terms• Skinny Client Control Protocol (SCCP) is a Cisco proprietary protocol

used between Cisco Call Manager and Cisco VOIP phones. Referred to as “Skinny” only work with the SIP protocol and an example of a vendor VOIP only control protocol.

• A Session Border Controller or SBC (IP to IP Gateway) is a device used in VOIP networks to allow control over the signaling and usually also the media streams involved in setting up, conducting, and tearing down calls. The are inserted into the signaling and/or media paths between calling and called parties in a VOIP call, predominantly those using the SIP, H.323, and MGCP call signaling protocols. Termed middle boxes between UAs and SIP servers.

• A Media Gateway acts as a translation unit between disparate telecommunications networks such as the PSTN and Next Generation Networks . Media Gateways enable multimedia communications across these disparate networks over multiple transport protocols such as ATM and IP for example.

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Almost Done…

• CODEC is a program capable of performing encoding and decoding on a digital data stream or signal. The word codec is actually just a combination of the words: “compressor - decompressor”. Common VOIP CODECS: G.711, G.729a, G.722 for example.

• An IP PBX is a business telephone system designed to deliver voice or video over a data network and interoperate with the normal Public Switched Telephone Network (PSTN). Cisco Call Manager, Avaya, Microsoft, are a few examples.

• A Soft Switch is a central device in a telecommunications network which connects telephone calls from one phone line to another, typically via the internet, entirely by means of software running on a general-purpose computer system that handles IP-to-IP phone calls. 2 types Class 4 and Class 5. SONUS and BroadSoft are examples.

• Jitter is the undesired deviation of frequency of successive pulses in electronics and telecommunications. Jitter is a significant, and usually a undesired factor in the design of almost all communications links.

UHGGGGG!

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SIP

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GenbandSBC

BroadsoftFeature Svr

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SAPP

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PSTN

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Basic Defined Elements in “Action”

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Brief History of VOIP and the

evolution of SIP with-in VOIP

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It all started in 1995 and VocalTec

• The history of VOIP shows that this technology started as far back as 1995 when a small company called VocalTec released, what was believed to be, the first internet phone software. This VOIP software was designed to run on a home PC and much like the PC phones used today, it utilized sound cards, microphones and speakers. The software was called "Internet Phone" and the hardware was called “Audio Transceiver” and used the H.323 protocol instead of the SIP protocol that is more dominant control protocol today.

• Anybody know what VocalTec is now most known for almost 20 years later?

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Control Protocol Evolution

Control Protocols: Around since the mid-90s•Used to set up and break down VOIP sessions (Similar to the ISDN-PRI D-channel in a TDM environment)•Types and different methodologies:

H.323 - older ITU standard (hard to program or use)

MGCP (Media Gateway Control Protocol) – mostly used in Hosted VOIP or IP Centrex – (never took off and Hosted has issues..)

SIP (Session Initiation Protocol) - Has become the de facto control protocol – (easy to program) SCCP helped / Beta vs VHS

SIPH.323 MGCP

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Data Stream Protocol History

After a VOIP session is setup using a Control Protocol (SIP) then a Data Stream Protocol invades •RTP (Realtime Transport Protocol) - Improves Quality of Service for VOIP data steams and used in VOIP today

• 2 RTP one way streams carry/enable the VOIP session (Similar to the ISDN-PRI B-channel in TDM Voice)

•RTCP (Realtime Transport Control Protocol) - used while the RTP Steam is running and piggy backs an RTP session to send summary reports back to sender

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“User Agents” perform a series of SIP Commands to talk

• Once again SIP is an Application Layer control (signaling) protocol for creating, modifying, and terminating sessions with one or more participants, known as User Agents.

• A series of SIP commands are used to accomplish the signaling tasks. Examples of these SIMPLE commands are:

• INVITE: Invites a user to a call• ACK: Acknowledgement is used to facilitate reliable

message exchange for INVITEs. • BYE: Terminates a connection between users

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THUS: SIP Session Call Flow – a closer look2 versions of the same SIP session with the left version providing more of the details. The

blue section shows the steps to setting up the session. The green section is the actual session using the two RTP streams and the Red section representing the breakdown steps

Setup

Session

Breakdown

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Deeper Dive on CODECs• References compression software to COmpress and

DECompress audio or video data streams to varying degree. Short for compress/decompress. CODECs can effect hardware and software (why there are many)

• Reduces the size of digital audio samples and video frames in order to:

• Speed up transmission• Save storage space

• Some CODECs discard bits that most people cannot hear or see for “bit saving” that effect quality levels

• Trunk Calls will have typically have less compressed CODECs while higher compression is used in the LANs behind the IP PBX.

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CODEC Specifics• G.711 is the default pulse code modulation (PCM) standard for Internet

Protocol (IP) private branch exchange (PBX) vendors, as well as for the public switched telephone network (PSTN). G.711 digitizes analog voice signals producing output at 64 kilobits per second (Kbps). Since the late 1970's G.711 has been the defacto standard in the telephony world for voice encoding as we moved into the digital world with fully digital phone switches, and moved away from analog phone exchanges. Since the mid 90's as VoIP has rapidly taken over in the telephony world and G.711 has still remained as the codec of choice.

• G.729 is an audio data compression algorithm for voice that compresses digital voice in packets of 10 milliseconds duration. Because of its low bandwidth requirements, G.729 is mostly used in Voice over Internet Protocol (VoIP) applications where bandwidth must be conserved.

• G.722 HD Voice and HD Audio have become the latest buzzwords in the VoIP (Voice Over Internet Protocol) market in the last year. They are all words to describe the same thing - wideband audio that delivers voice calls using VoIP with audio quality that is greatly superior that of a regular landline or mobile phone call.

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CODECs

G.711 is the default CODEC for IP PBX vendors.and the PSTN

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CODEC Misconceptions

• G.711 is roughly 100K (87.2K) per call so a DS1 or 1.5m can handle 15 simultaneous calls.

• G.729 is roughly 40K (31.2K) per so a DS1 of 1.5m can of IP can handle 35 simultaneous calls

Obviously G.729 can save you money from the Vendor trunk side being less bandwidth is needed for more calls but if the design is off degradation, echo, dropped calls, etc can develop and VOIP/SIP can take the blame when really it just the CODEC. How? Remember when I said the PSTN is G.711? Scenario..

What about Jitter and SIP/VOIP?

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JITTER

• Jitter may be caused by electromagnetic interference (EMI) and crosstalk with carriers of other signals. Jitter can cause introduce clicks or other undesired effects in audio signals, and loss of transmitted data between network devices. The amount of tolerable jitter depends on the affected application.

• Typically VOIP and SIP needs to operate with nothing more than 5ms of jitter at a max or again echo and degradation will occur.

• Your SIP provider/vendor is very key to your success with SIP service being your providers network is what connects your SIP service for completion. Is your vendors network a shared or dedicated service? What is the latency on the network either layer 2 or layer3? Is a fiber based service or copper? Many more – YOUR PROVIDER is KEY!

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Ok - a little bit of bun

Month/Year Packet Delivery % Latency (ms) Jitter (ms)

June-2011 100.00 38.87 0.03

May -2011 100.00 39.52 0.04

April-2011 100.00 39.71 0.04

March-2011 100.00 39.79 0.04

February-2011 100.00 40.1 0.04

January-2011 100.00 40.14 0.04

Dec-2010 100.00 39.64 0.04

Nov-2010 100.00 40.14 0.05

Oct-2010 100.00 39.22 0.06

Sept-2010 100.00 39.64 0.05

Aug-2010 100.00 39.65 0.04

July-2010 100.00 40.21 0.07

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How about SIP & Fax MachinesEarly on SIP had and developed real fax issues mostly because G.729

was being pushed to early. Issues especially developed with FAX Servers:

• Set the transmission speed to 9600 (BAUD Rate) • Use only G.711 with any compression like G.729 • Set the Resolution to Standard.

Three forms of fax over IP networking:

• Realtime fax using the T.38 protocol and T.38 based fax gateway devices installed on the IP network.

• Internet fax - Also known as T.37. The ITU standard for sending a fax-image file via e-mail to the intended recipient of a fax.

• VoIP based fax - Also known as G.711 pass through - This is where the fax call is carried in a VoIP call encoded as audio. Most Vendors only support this type of fax.

.

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What about 911 Service?

Companies like Vonage and residential type Vendors and Providers really hurt 911 and VOIP reputation early on. Today E911 issues are solved with advances in 911 service and PS/ALI (private switch/automatic location identifier) with the PSAP itself to give the ability for multiple emergency response locations per trunk group.

VTN 911 which uses Foreign Rate Centers is typically not supported at the remote location by most Vendors

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Why SIP/VOIP?

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VOIP/SIP Value• “One Wire to the Desktop” – Converged Network Infrastructure

• Common cabling to the desktop• Saves 50%

• “Toll Bypass” - Site-to-Site Communications

• Eliminate Moves, Adds & Change Charges• Companies typically spend $119/MAC• 0.87 MACs/employee/year

• Portability & Telework

• Features, Features, Features

• Key To Disaster Recovery Plans• Pick up phones and deploy to new locations

Are you using or planning to use IP telephony?

19%

13%

3%

3%

62%

Currently RunningTrialPlan To Implement1-2 YearsPlan To Implement6-12 MonthsPlan To Implement0-6 MonthsCurrently Using

Is the quality of IP telephony holding you back from further application?

35%

65%

Yes

No

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SIP Value Propositions

• Versatility SIP can be used for telephony, notification services, location services, collaboration, chat and conferencing

• Extensibility SIP’s internal structure makes it easy to add new primitives — i.e. signaling protocol elements without

disrupting existing primitives.

• Multimedia at the coreSIP natively takes into account audio, video and text sessions.

• Mobility across IP networksA registration and location mechanism enables mobility of endpoints over various IP networks.

• IT-friendlySIP leverages other existing, well-established Internet protocols, such as Domain Name System (DNS) and

Simple Mail Transfer Protocol (SMTP). SIP also leverages Internet Protocol Security (IPSec) to provide session encryption and security.

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SIP and IP PBX Market - (lettuce)

• The VOIP service market continues to grow:• $34.8 billion in 2008 • $49.8 billion in 2010 • $74.5 billion expected by 2015

• SIP trunking had 143% revenue growth in 2010 alone.

SIP is becoming a key product line for Vendors

& Vendors will spend money on Development

Source: IDC, August 2009 Source: IDC, August 2010

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PSTN Sunset Coming! SIP will Grow!• A Technical Advisory Council (TAC) recommended on June

29, 2011 to the FCC they set a “date certain” for the sunset of the PSTN.

• When will the PSTN “end”? A recent study by the National Center for Health Statistics says it all.

As of My 2010:• 23% of respondents lived in a mobile-only household

• 37% of adults in the 18-24 and 30-34 age groups• Only 6% of the US population will still be served by the PSTN

(defined as TDM access line service) by the end of 2018

• What will replace the PSTN?• Some future technology?• Cell (mobile)• VOIP/SIP has the lead

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SIP $ Misconceptions?VOIP and SIP calls are free from 800 charges?NOT

VOIP and SIP calls are free from LD charges?NOT

SIP will save me hardware cost with Softphone usage?NOT

SIP call quality is not up to par and could cost my company’s image?NOT

SIP will save me hardware cost with less Voice TDM cards to buy for my legacy TDM PBX?

TRUE

SIP will save me DR downtime cost with phone mobility?TRUE

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What does the Future Hold?

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Today’s Features

Users are attracted to feature sets:

• Advanced User Interface • Find Me Follow Me• Visual Voicemail• Caller ID Customization• Voicemail to Email • Inbound Call Description • Announcement Interface • Call-out • Call Pickup• System Diagnostics • Multi vendor Phone Options • Analog Phone Support • BYO Phones

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Tomorrow: More Features (tomato and mayo)

• Cause Code Routing/SIP Responses/Crank Back Mapping“SIP with SIZZLE”

Manipulating SoftSwitch response codes for call priority!!!

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The Future of SIP

• SIP history is short but growth is dramatic• Three major trends driving large enterprise

communications: 1. Globalization (no boundaries)

2. Unified communication solutions for all (new generation of users)

3. Interweaving of communications applications

• SIP versatility is a key to all three trends• Standard still evolving and interoperability improving

SIP will become ubiquitous in large enterprise networks within the next 2 to 5 years.

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Thank You

Jon MurphySr. Network Application Engineer

jon.murphy@twtelecom.com(614) 255-2132 (office)(614) 313-6925 (cell)

Jon MurphySr. Network Application Engineer

jon.murphy@twtelecom.com(614) 255-2132 (office)(614) 313-6925 (cell)

Questions and AnswersQuestions and Answers