January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking 101 Presenters: Pete Sandstrom, CTO BandTel Janne Magnusson, Director Operations Ingate

January 23-26, 2007• Ft. Lauderdale, Florida

SIP Trunking 101Presenters:

Pete Sandstrom, CTO BandTelJanne Magnusson, Director Operations Ingate


SIP Trunking 101 Session Overview

1. Business Opportunity

2. What is Signaling

3. The SIP & SIP Trunking, Value Prop

4. The Role of the ITSP


1. Business Opportunity is Primed

• IP-PBX Market on track to exceed $10 billion

(Synergy Research)

• By 2008, 55% of all Enterprise

Communications Systems will be IP-enabled

(Frost and Sullivan) which means the

majority of PBXs will be

SIP-compatible/ready

• VoIP Service Providers gaining enormous

traction in marketplace


2. What is Signaling?

Since SIP/SIP Trunking is a standard for signaling, let’s

understand what signaling is and the many forms of Voice

Over Internet Protocol (VoIP). The 4 major standards-

based signaling systems are:

1) H.248 Media Gateway and Control (MEGACO)–

an International Telecommunications Union (ITU)

(www.itu.it) standard. H.248 originated from the Public

Switched Telephone Network (PSTN) using Signaling

System 7 (SS7) with dumb telephones.



2) H.323 is an ITU standard and the logical

progression from circuit-switched to packet-

switched telephone call processing including

video services. H.323 networks consist of Call

Processing servers, Media Gateways (MG) and

GateKeepers.



3) Media Gateway Controller Protocol (MGCP) is a device

control protocol developed by Internet Engineering Task

Force (IETF) designed to control devices such as Media

Gateways (MG) and Integrated Access Devices (IAD).

The difference between MGCP and other multimedia control

protocol systems is that MGCP allows the endpoints in the

network to control the communication session.



4) Session Initiation Protocol (SIP) - an Internet

Engineering Task Force (IETF) signaling protocol

for internet conferencing, telephony, presence,

events notification (emergency calling) and instant

messaging.

For complete details on SIP, go to www.ietf.org

RFC-3261 or http://www.ietf.org/rfc/rfc3261.txt


A. Bell – The Beginning of Signaling

• Dumb phones & Dumb network• Signaling via Tip & Ring – Loop Start – Ground Start


• Dumb phones & Smart network• Signaling via inband (T-1) or out-of-band (ISDN)

Ma Bell – Signaling System 7 (SS7)Designed in the 1980s, deployed in the 1990s


• Smart phones & Smart applications SIP network• Signaling is via SIP messages sent via UDP/IP

No Bell – Internet Adds SIP


3. What is Session Initiation Protocol (SIP)?

• SIP is a telephony signaling protocol that is used to establish a "communications session or connection" such as a telephone call, conference call or other type of communications on an Internet Protocol (IP) network.

• SIP is a request-response protocol that operates like a "communications browser" protocol such as Hypertext Transfer Protocol (HTTP).

• SIP is the communications equivalent to such internet protocols such as HTTP and Simple Mail Transfer Protocol (SMTP).


SIP “Invites” Callers Together


SIP is a Fully-Featured Protocol


What is SIP Trunking?

• SIP Trunk is a network interface device that

recognizes SIP signals (signaling) and can

process these signals to other SIP devices.

• In other words, a SIP Trunk is an expandable or

“virtual” digital data transmission link with SIP

protocol signaling interfaces that recognizes SIP

signaling and SIP media using Real-Time Protocol

(RTP) for media transmission (voice, rich media).


SIP Trunking Applications– Local and inbound calling

– G.711 and G.729a Codecs

– Re-Invite support across multiple locations

– Inbound load balancing options across multiple locations

– Disaster recovery options

– Directory Assistance

– Enhanced 911 services access

– 411

– Directory Listing

– Operator assisted dialing

– Local Number Porting Management


SIP Trunking Value Proposition SIP has become “the” defacto standard for IP communications:

• SIP offers a set of formal rules that provides the basic signals used to initiate, manage, and terminate IP communications sessions.

• SIP goes beyond voice to support true multimedia communications/convergence – i.e., voice, data, video

• SIP embraces and supports enhanced communications capabilities, such as presence and mobility applications

• Reduces toll charges from SIP origination/termination services

to the PSTN for external long-distance and local access calls

• Optimizes Utilization of Enterprise IP Pipes


SIP Supports Legacy Infrastructures

• SIP supports for both Legacy and New IP-based Enterprise Communications Systems

• SIP offers a migration path while retaining investment in legacy equipment while bringing the unique benefits of VoIP

• SIP offers:– Reduced Transport Costs– Enhanced Services: DID numbers, call transfer

between headquarters/branch offices, real-time access to billing and usage


IP-PBXs Migrate PBXs – ITSPs Emerge

ITSP

PTSN

IP

PBX

SIP Services

GW

SAFW

SIP-Aware FireWall (SAFW)


ITSP Do More

• Public vs. Private IP Pipe• ITSP should be able to work in both

domains

- Public Internet

- Private – Virtual Private Network (VPN)

- Both can be completely secure, but VPN costs more


SIP Over Public Internet or VPN






Security and Reliability

• Need to see that the CPE LAN is correctly designed for VoIP

• CPE protection- a SIP Aware Firewall (SAFW) is needed allowing Layer 5 (L5) security (i.e. no L2 pinholes)

• ITSP Message Digest 5 (MD5-digital encryption coding) is minimal requirement IP authentication is needed

• ITSP should split media and signaling to different redundant locations

• ITSP must have secure POPs that can fend off all outside attacks, including source-route bridging spoofing detection.


4. The Role of the ITSP

• SIP User Agent is Plug-and-Play with ITSP

(Interoperability Program)

• ITSP Architecture Black Box to SIP User Agent

• Core ITSP infrastructure is redundant and resilient

• Rich diversity of PSTN carriers and routes to

global destinations


What Does an ITSP Look Like


ITSPs “Peer” For Customer Performance


ITSP “Peering” For Total SIP Solution

ISP

IP

PBX


SIP Trunking 101 Summary

• SIP Trunking brings “plug-and-play” interoperability for existing

PBX systems and hosted VoIP systems

• SIP Trunking simplifies deployment and support chores for

distributors and channel partners

• SIP Trunking brings offers new features/services in the form of

“open” applications development

• SIP Trunking provides geographical unification of the enterprise

• SIP Trunking brings enormous cost savings to customers




SIP Trunks

Internet

IP-PBX

Firewall

Service Provider

• Important to have a reliable and well dimensioned network

– Consider delay and QoS

• The firewall is the demarcation point for the enterprise network

• Secure privacy and prevent eavesdropping

Communication on the LANSIP Trunks

Internet

IP-PBX

Firewall

Service Provider


SIP Trunks

Internet

IP-PBX

Firewall

Service Provider

• Important to have a reliable and high quality Internet connection

– Consider delay to ITSP– Of your connection QoS

(voice should have priority)

• Voice travels over public Internet (as e-mail)

• Possible to secure privacy and prevent eavesdropping

Communication outside the LANSIP Trunks

Internet

IP-PBX

Firewall

Service Provider



Questions?


About Ingate

• Formed 2001– Firewall technology from Cendio Systems

• Appliance firewalls since 1994– Capital and SIP technology from Intertex Data AB

• Began SIP development in 1998

• Released the worlds first SIP capable Firewall in 2001

• Located in Stockholm and Linköping, Sweden with a subsidiary, Ingate Systems Inc., based in Hollis, NH.

• Confirmed IP-PBX interoperability:3Com, Asterisk, Avaya, Broadsoft, Cisco Call Manager, Ericsson MX-One, Mitel, Pingtel, SER, Shoretel, Sphere, Swyx, Zultys

• Confirmed carrier interoperability:Bandtel, Broadband.com, Cbeyond, Global Crossing, IP-Only, O1, RNKTel, Tele2, VoEx


About BandTel

• Headquartered in Newport Beach, California, BandTel is a leading worldwide provider of SIP Trunking services.

• BandTel is dedicated to ensuring its customers and partners alike have access to the most reliable, end-to-end VoIP service available on the market today.

• BandTel’s N-Plus™ network architecture is designed to solve the throughput and redundancy problems on high-capacity SIP-based networks and eliminate any single point of failure.

• BandTel continues to develop strong partnerships with leading carriers and telecommunications companies, including Global Crossing, XO Communications, Level 3, Qwest Communications, Verizon Business, and Primus.

Documents

January 23-26, 2007 Ft. Lauderdale, Florida SIP Trunking 101 Presenters: Pete Sandstrom, CTO BandTel Janne Magnusson, Director Operations Ingate