Transcript Slide 1

SIP – Yesterday, Today, &
Tomorrow
Jon Murphy
Sr. Network Application Engineer
tw telecom
Introduction
Jon Murphy
Sr. Network Application Engineer
tw telecom
[email protected]
(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…
Warning No “Commercials”
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•
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!
Start with a “Knowledge Foundation”
5
•
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
School’s still in: More Basic VOIP Terms
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•
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.
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.
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UHGGGGG!
Basic Defined Elements in “Action”
FortisVox
EMS
Customer
Premise
CISCO IP PHONE
7960
1
2
ABC
messages
3
directories
i
DEF
services
4
5
GHI
JKL
7
settings
6
MNO
8
9
PQRS
TUV
WXYZ
*
0
#
OPER
Ethernet
Switch
IP PBX
Genband
SBC
BS Media
Server
FortisVox
eSBC
SIP
CISCO IP PHONE
7960
Trunk
1
2
ABC
3
messages
7
5
JKL
HAGG
SAPP
directories
i
DEF
services
4
GHI
settings
6
MNO
8
9
PQRS
TUV
WXYZ
*
0
#
OPER
Sonus
GSX
PSTN
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Sonus
PSX
Broadsoft
Feature Svr
tw telecom
IP Core
Brief History of VOIP and the
evolution of SIP with-in VOIP
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It all started in 1995 and VocalTec
•
•
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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?
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
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H.323
MGCP
SIP
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 look
2 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
Codec Information
Codec
Codec Mean
Codec & Bit
Sample Sample Opinion
Rate (Kbps)
Size
Interval Score
(Bytes)
(ms)
(MOS)
Bandwidth Calculations
Voice
Packets
Voice
Bandwidth
Payload
Per
Payload
Ethernet
Size
Second
Size (ms)
(Kbps)
(Bytes)
(PPS)
G.711 (64 Kbps) 80 Bytes
10 ms
4.1
160 Bytes
20 ms
50
87.2 Kbps
G.729 (8 Kbps)
10 Bytes
10 ms
3.92
20 Bytes
20 ms
50
31.2 Kbps
G.723.1 (6.3
Kbps)
24 Bytes
30 ms
3.9
24 Bytes
30 ms
33.3
21.9 Kbps
G.726 (32 Kbps) 20 Bytes
5 ms
3.85
80 Bytes
20 ms
50
55.2 Kbps
G.728 (16 Kbps) 10 Bytes
5 ms
3.61
60 Bytes
30 ms
33.3
31.5 Kbps
10 ms
4.13
160 Bytes
20 ms
50
87.2 Kbps
G722_64k(64
Kbps)
80 Bytes
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
20
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
How about SIP & Fax Machines
Early 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 faximage 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%
Are you using or planning to use IP
telephony?
19%
“Toll Bypass” - Site-to-Site Communications
13%
62%
•
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
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3%
3%
Currently Running
Trial
Plan To Implement
1-2 Years
Plan To Implement
6-12 Months
Plan To Implement
0-6 Months
Currently Using
Is the quality of IP telephony holding you
back from further application?
35%
Yes
No
65%
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 core
SIP natively takes into account audio, video and text sessions.
• Mobility across IP networks
A registration and location mechanism enables mobility of endpoints over various IP
networks.
• IT-friendly
SIP 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
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Source: IDC, August 2010
Source: IDC, August 2009
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:
•
•
•
•
•
•
•
•
•
•
•
•
•
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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
Tomorrow: More Features (tomato and mayo)
•
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Cause Code Routing/SIP Responses/Crank Back Mapping
“SIP with SIZZLE”
Manipulating SoftSwitch response codes for call priority!!!
The Future of SIP
• SIP history is short but growth is dramatic
• Three major trends driving large enterprise
communications:
Globalization (no boundaries)
2. Unified communication solutions for all
(new generation of users)
3. Interweaving of communications applications
1.
• 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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Questions and Answers
Thank You
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Jon Murphy
Sr. Network Application Engineer
[email protected]
(614) 255-2132 (office)
(614) 313-6925 (cell)