SIP Trunking overview

Presented by:
John Downing, B.Eng, MBA, P.Eng

John Downing
◦ co-founder of TrainingCity.
◦ VoIP Training Development Lead.
◦ VoIP & SIP Consultant to Telecom & Enterprise
Clients.
◦ John@TrainingCity.com
© TrainingCity 2012 1.2

In this seminar we will explore:
1. What is SIP Trunking?
2. The importance of SIP: Session Initiation Protocol
3. Why customers are shifting to VoIP
4. How to Sell SIP Trunking
5. The Marketplace: What’s out there:
1. Legacy Trunks
2. Other Carriers / ISPs
3. Internet Telephony Service Providers
6. Questions & (hopefully) Answers!

What exactly is “S”IP Trunking?
◦ Connect the PBX to the carrier using TCP/IP data
networking instead of legacy TDM (Time Division
Multiplexing) Trunk technologies.
◦ SIP Trunking extends the concept to include ‘end to end’
connectivity for VoIP calls through the data network.
◦ More than just a replacement for the T-1/ISDN PRI
Trunk. SIP Trunking is an “end to end” solution delivered
over an IP Network.

What exactly is an IP Trunk?
◦ TCP/IP Bandwidth + SIP Signalling
◦ Bandwidth for Voice & Video transport
◦ Signalling to interconnect client VoIP infrastructure
with:
Other Carrier SIP Trunking Clients
Carrier Provided PSTN Gateways

T-1 & ISDN are inefficient:
◦ Traffic Engineering for traditional PBX requires the addition
of extra T-1 or PRI in groups of 24/23 channels.
◦ Bandwidth for SIP Trunking can be made available in any
increment.
◦ SIP Trunking is more than just a replacement for TDM
Trunks. It is a solution for “end to end” VoIP connectivity &
integral to all future Carrier Value Added Service offerings.
◦ Irony is that SIP, a TCP/IP Protocol, may finally achieve the
carrier’s old SS7 dream of an “Intelligent Network” offering.

One IP PBX
PSTN Gateways with TDM Trunks distributed
geographically.

SIP Trunking can scale more easily than
legacy ISDN PRI Trunks.
The Physical Layer of SIP Trunking is “any” available
bandwidth, not limited to incremental additions of
PSTN circuits and associated PBX hardware.
If a client needs more VoIP calling capability at a
specific site, add additional bandwidth to access
router.

Use existing WAN for interconnectivity of
Offices:
◦ A multi-site corporate client can optimize its supply
chain by eliminating contractual arrangements with
multiple PSTN access providers.
◦ Use existing WAN to backhaul VoIP calls to Rogers
supplied PSTN Gateways.

Hard to get head around idea that a SIP Trunk
extends through the WAN. For a client this
means a SIP Trunk could connect their
corporate LAN to any PSTN Gateway in the
Carrier WAN, or another client of the Carrier.
Next step is to offer Trusted SIP Peering
arrangements with other carriers.

SIP defined in RFC 3261
◦ Establish & manage multimedia sessions.
◦ Client / Server : Request / Response Architecture
◦ Similar to HTTP:
SIP/UDP/IP/Layer 2
May use Session Description Protocol
SDP/SIP/UDP/IP/Layer 2
◦ Exchange SIP messages between end points

SIP dominates Enterprise and Carrier VoIP,
Video, & Multimedia deployments.
All equipment vendors actively support
standard.
Other VoIP Signaling standards are obsolete:
◦ H.323
◦ MGCP
◦ Vendor Specific such as Cisco SCCP
◦ IAX2 (Possible exception: Asterisk specific and ETF
std)

SIP is part of TCP/IP.
Considered an Application Layer Protocol in
TCP/IP just let HTTP & FTP. Works the same
way.
SIP messages use regular TCP/IP packets for
transport from IP Phone to IP PBX.
Takes advantage of all IP Security.

SIP Does not transmit the voice directly.
SIP “sets up” the Voice Over IP packets for
transmission. Voice is sent using a protocol
called Real Time Protocol (RTP).

VoIP consists of two sets of IP Packets:
◦ Real Time Protocol (RTP) packets
Often referred to as an “RTP Stream” because the voice
(and video) is actually carried in RTP.
◦ Session Initiation Protocol (SIP) Packets
SIP Packets, or SIP messages as they are called, setup
and teardown the RTP Streams.

Server sends back a response message
CALLEE
Client sends a request message
5.20
User
Agent
Client
User
Agent
Server
User
Agent
Client
User
Agent
Server
CALLER
INVITE
200 OK

VoIP calls consist of SIP Messages
Signalling and Real Time Protocol
Packets to transport the voice.
As only a few SIP messages are needed
to setup most VoIP calls, the bulk of the
Bandwidth requirement is for the RTP
Streams

Item
• Standard
• Relationship
• Servers
SIP
• IETF
• Peer to Peer &
Client Server
• Proxy, Registrar,
Location, Redirect

Item
•Media Capabilities
Exchange
•Security
•Quality of Service
SIP
•Uses Session
Description Protocol
•ITEF Standards, TLS,
IPSec, etc.
•IETF Standards,
DiffServ, RSVP, etc.

SIP is the signalling standard for all Voice &
Video over IP.
SIP is unlikely to be replaced any time soon.
Lots of innovation occurring around SIP.
Clients need to prepare for a world where
TDM Legacy voice is no longer supported.

PSTN (Public Switched Telephone Network):
◦ Transport of Voice using concept of Circuits
TDM - Time Division Multiplexing examples such as T-1,
ISDN PRI.
Analog “CO” circuits.
◦ Signalling:
Various access technology standards build around core
network signalling based on SS7.
VoIP:
◦ Transport over IP (Packet) networks
Layer 2/IP/UDP/RTP/Codec
◦ Signalling using SIP – Session Initiation Protocol
Layer 2/IP/UDP/SIP

VoIP Challenges include:
1. Meeting all the specific requirements of “making
a phone call”:
IT networks designed for “best effort”, shared
infrastructure, & optimized to minimize cost while
providing reasonable service.
2. Regulatory requirements:
Calling 911
law enforcement intercepts
Billing & Tariffs

Challenges:
◦ Multi-vendor legacy PBX support
Geographic & Multi-vendor specific PBX support
demands.
◦ Carrier/Supplier Management
Need separate carrier agreements for many diverse
locations.
◦ Physical Plant
Maintaining facilities for large, complex PBX Hardware.

Core components of Enterprise VoIP include:
◦ IP Phones
◦ IP PBX
◦ PSTN Gateway(s)
◦ Session Border Controller
In carrier infrastructure the focus is on
transport. Rather than an IP PBX the carrier
may deploy a “soft switch” server.

Two basic types of IP PBXs in the market
today:
◦ Legacy PBXs that are also VoIP enabled.
◦ “Pure” VoIP IP PBX.

Phones connect to LAN using Ethernet +
TCP/IP.
PBX becomes an “IP” PBX. Also uses Ethernet
+ TCP/IP.
No need for PSTN trunks when calling another
VoIP Phone in the network.
PSTN trunks can be accessed using a
VoIP/PSTN Gateway.

IP PBX replicates functionality of 1990s
proprietary TDM PBX:
◦ All the features of a PBX replicated in VoIP:
Call forward
Do Not Disturb
Transfer, Hold, etc, etc, etc.
Now just another Server on the network:
◦ Add on functionality such as Voice Mail can be
“outsourced” to a dedicated “Voice Mail Server”.
◦ Same for IVR (Interactive Voice Response) or any
other function.

The PSTN Gateway device is designed to
convert:
◦ VoIP transport to/from Circuit transport
Layer 2/IP/UDP/RTP/Codec to/from TDM Codec
◦ SIP signals to SS7 based PSTN signals
Layer 2/IP/UDP/SIP to/from Q.Sig or ISUP based SS7
The PSTN Gateway may also have to perform
codec conversion:
◦ PSTN T-1 interface requires G.711 μ-Law but VoIP
transport sent voice encoded in G.729.

SBC provides “Web Proxy Server” functionality for VoIP & SIP.
Note that RTP Streams and SIP Dialogs both connect through the SBC.

SIP Trunk implies much more than just “over T-1”.
Some clients may think this is all SIP trunking offers.

Corporations may opt for Private PSTN Gateways. “SIP Trunking”
on their WAN.
Still can offer access to RBS PSTN Gateways & other QoS
management services.

Client with several locations can use SIP
Trunking over an IP WAN to interconnect their
IP PBXs.
Provide incoming and outgoing calls to the
PSTN through the supplied SIP Trunks. Client
no longer requires T-1 or PRI trunks at every
location, just access to the WAN.

Solution with SIP Trunking must include:
◦ Provide all basic call services, LD, DID, Toll Free,
CLID, etc.
◦ Shared bandwidth usage across the WAN. Setup
SLAs to permit client to scale Quality Assured VoIP
bandwidth across multiple WAN access locations.
◦ E-911 support.
◦ Redundancy & backup.
◦ Security
Private Network
VPN connectivity for VoIP & SIP traffic
Secure codecs, SIPs, SRTP, should all be supported.

Client can trust Rogers to provide secure access to VoIP

Yes it works if configured properly.
Option One:
◦ Route call to 911 contact centre, tell operator
location, then get routed to correct Public Safety
Answering Point (PSAP).
Option Two:
◦ Deploy at least one local PSTN gateway and
associated PSTN trunk at each location, configure
SIP to send all 911 calls to locally (LAN) deployed
PSTN Gateway.

Ideas to Start Client Conversations:
◦ SIP is the Future.
No one is still developing old TDM equipment.
Sooner or later old TDM PBX equipment will reach “end of life”. Get out
in front of impending requirements.
◦ All sorts of SIP enabled Unified Communications solutions are out
there.
Opportunities to align business processes with productive SIP enabled
solutions are endless:
Unified Communications: email/vmail/messaging/social media.
Customer interaction solutions.
Contact Centre integration (infinitely easier & better than old TDM
solutions from the 1990s!)

Important to educate client on idea that SIP
Trunking is more than just a replacement for old
TDM access trunks.
SIP Trunks extend IP PBX call control into the
WAN.
SIP Trunks allow client to access RBS PSTN
Gateways… AND RBS Services!
◦ Amazing opportunity for RBS to (finally) start offering
enterprise grade managed voice/video/productivity
solution(s).

Clients will be looking for a wide range of
capabilities including:
◦ Protocol Support
◦ Service Providers technical capabilities
◦ Cost
◦ Security
◦ Availability
◦ Interface options & protocol support
◦ Network geographically diversity.
◦ Path to Unified Communications

Internet Telephony Service Providers range
from Vonage to “Basement BBS Guys”.
◦ They scale with the Internet, but so does RBS.
◦ Hard for enterprise clients (& IP PBX vendors) to
know who to trust.
Rogers is a trusted WAN partner with extraordinary
security & reliability expertise.
ITSP is “fly by night”, accident waiting to happen with
confidential information.

Beyond the SIP Trunk to PSTN Gateway, think
about managed SIP Services:
◦ Voice Mail Servers
◦ Contact /Social Media Centre scaling
◦ Content Server integration
Security, Security, Security
◦ You’ve got private WAN capability
◦ You can secure VoIP traffic that needs to flow into
The Internet through Session Border Controllers &
Application Gateways.

Partner with other geographically diverse
Carriers to offer “Trusted SIP Trunks” to other
jurisdictions.
◦ Offer clients secure SIP trunking solution that
reaches USA & global locations.

Small to Medium Enterprise clients benefit:
◦ Combine voice & data network access.
◦ Eliminate onsite TDM trunks to PSTN by using
Gateways.
◦ Reduce capital costs.
◦ Take advantage of SIP based productivity
enhancement features associated with IP PBXs such
as:
Unified Communications
Follow me features
Etc.

Clients of RBS could:
◦ Avoid the requirement to maintain multiple-facility
TDM solutions.
◦ Establish a point of presence without having to
build local facilities.
◦ Point E.164 numbers to any trunk group/IP
address/URL
◦ Provision and Scale On Demand
◦ Port over existing PSTN DID numbers

Not just PRI replacement
◦ Multiple locations served from a single IP PBX
Capital and expense savings with converged solutions
Support features like single number service
Multi-location services
SIP trunks more flexible and efficient
◦ Part of business case to support UC deployment

SIP Trunking is not just a “trunk replacement”
It is migration from TDM to Voice over the
data network.
Is equipment SIP compliant?
Professional services to perform:
◦ Protocol check
◦ Project Management
◦ Capacity planning

Voice networking still dominated by TDM
infrastructure.
SIP Trunking is an enabling technology
◦ Brings UC to organizations
How to evolve Voice infrastructure to SIP
Trunking
How to choose a SIP Trunking service
provider.

Breaks down into three main groups:
◦ Internet Telephony Service Providers (ITSP)
Skype/Microsoft, Vonage, Google, “no name” small
players
◦ Traditional Carriers
Bell, Verizon
◦ Unexpected entrants
Akamai?

ITSPs range from major players to “one guy in
his parent’s basement:
◦ Can’t dismiss either, lots of great companies
started in garages.
◦ Major players Vonage & Skype
◦ Hard to dismiss the 800,000,000,000 Gorilla called
“Google”.
◦ Mid sized players.
◦ Small ITSPs that offer all sorts of quasi-legal VoIP
services globally.

Geographical access still defines much SIP
Carrier competition.
Need ability to offer corporate clients physical
access to carrier WAN.
Regulatory issues remain important, but not
as strict as legacy PSTN.
Bottom line, usual suspects… for now.

Bell offers SIP services to large enterprise
customers in Ontario & Quebec.
Expanding in West
Rolling out SIP to SMEs over past year.
Reputation for excellent in SIP knowledge.
Working to “change the culture” of the company
to VoIP & SIP through effective Training &
Education programs.

Must have MPLS VPN (Layer 3 or Layer 2 Ethernet)
connectivity to access SIP Trunking.
Voice Quality
◦ Provisioning the right QoS in the Data Network, engineer
for voice with strict considerations for packet loss, jitter,
& latenancy.
Bell offers clients the option to access their traffic
status, usage of SIP trunking bandwidth.
Service Level Objectives:
◦ Mean Time to Repair 4 hours
◦ Availability 99.9%

Long Distance pricing
◦ No more long distance for calls internal on WAN
◦ Make calls from IP PBX to any rate centre with trunk
group in place
◦ Calls on trunk groups that are not local to the
destination charged according to rate plan.

Verizon offers a wholesale SIP Gateway
service:
◦ Uses IPSec to transmit SIP & RTP (VoIP calls)
between client carrier network & Verizon network.
◦ Client can access multiple Verizon PSTN Gateways.
◦ Termination of both local & long distance to/from
Verizon SIP/VoIP network to PSTN.
◦ Includes support for FTNA (Foreign Telephone
Number Assignment capability.

Verizon offers a wholesale SIP Gateway
service:
◦ Pricing models vary with carrier client requirements.
◦ Monthly recurring charges
◦ Non-recurring charges for specific services
◦ Metered per minute pricing
◦ Domestic usage rates based on terminating Local
Access & Transport Area (LATA) or classification of
the terminating Local Exchange Carrier (LEC) based
on the dialed NPA-NXX.

◦ As long as a proper mixture of Interstate and Intrastate rates can be
maintained, discounts are available for certain call jurisdictions.
◦ Interstate- and intrastate-defined traffic will rate at a six-second
minimum/six-second increment.
◦ The international rate structure is based on the terminating location
(landline and wireless termination).
◦ The extended domestic traffic will rate at a six-second minimum/six-second
increment.
◦ The international traffic will rate at a 30-second minimum/six-second
increment, except for Mexico, which will rate at a 60-second
minimum/60-second increment.
◦ http://www22.verizon.com/wholesale/solutions/solution/sip%2Bgateway%
2Bservice.html

Feel free to email questions to John or Kerry
later if desired.
If you are interested in receiving email
updates of the slide presentation and SIP
Trunking resources please see Kerry before
leaving to be added to our email distribution
list.

Call or email us anytime with questions:
SIP Technical Questions:
◦ John Downing
◦ John@TrainingCity.com
◦ 613-435-1170
Professional Services related to SIP
Deployments & Training Requirements:
◦ Kerry Parnell
◦ Kerry@TrainingCity.com
◦ 613-799-6750

Review of equipment configurations
Bandwidth requirements analysis
Training:
◦ Course 784: VoIP Hands ON Workshop – 3 Days
◦ Course 2004: SIP Hands ON Workshop – 2 Days

TrainingCity Classes available on-site and as
Public Deliveries:
◦ Course 2004: SIP Hands ON Workshop
◦ Course 801: IPv6 Fundamentals
◦ Course 802: DNSSEC Hands ON
◦ Course 784: VoIP Hands ON Workshop
www.TrainingCity.com

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