SIP Trunking Seminar

SIP TrunkingSeminar

Can Your PBX do That?October 2006

2Pingtel ConfidentialMay 3, 2005

Why is SIP Trunking so difficult ?

• Well, is it really?

-Firewall vendors still don’t know SIP

-All the trouble because of NAT

-Lines vs. Trunks (do you need DID?)

-Security (to register or not to register?)

-Is it really cheaper?

-What about voice quality?

-What else can I do with this?

We understand SIP well and wrote big parts of the standard


The Good News: It Really Works

• A SIP Proxy based IP PBX does inter-domain routing by default

• Use a specialized SBC or SIP aware Firewall(being able to transfer calls is important)

• Insist on certified offerings or bundles

• Remote workers and branch offices can now be connected too


Lines vs. Trunks

B2BUA

SIP

SIP

SIP

SIP

SIP

SIP

ITSP

508-345-1123

508-345-6532

508-345-9987

Poor Man’s Trunking – Good for small deployments

Holes

• No DIDs• Security through

registration (like phones)• Holes in your firewall• STUN keep-alive traffic

• No DIDs• Security through

registration (like phones)• Holes in your firewall• STUN keep-alive traffic


Deployment Example

Ingate SIParatorsipX Media &Config Server

sipX HA Call Control Server

Load Balance &Fail Over

• DIDs• Different possibilities for security (ACL,

TLS, SIPConnect)• No holes in your firewall – stateful

inspection of SIP traffic• Fail-over in case route is unavailable or

busy – fully automatic• Works with complex multi-party call flows

• DIDs• Different possibilities for security (ACL,

TLS, SIPConnect)• No holes in your firewall – stateful

inspection of SIP traffic• Fail-over in case route is unavailable or

busy – fully automatic• Works with complex multi-party call flows


Is it really cheaper?

• PSTN gateways are expensive:

-Single span T1/E1 $3,200

-Dual span T1/E1 $4,380

-8 span T1/E1 $15,500

• Who does the best job on least cost routing?

• SIP aware firewall or SBC needed anyway for remote workers and branch office interconnection

• SIP Trunking can save up to 40% on CAPEX


What About Voice Quality?

• Depends on the quality of your data connection

• Fewer format conversions improves quality

• Polycom’s HD Voice supported for on-net calls only

• Requires proper network engineering: VLAN, QoS, TOS

• Usually results in higher voice quality


What else can be done with this?

• Phase 1: Call Origination and termination to and from the PSTN using a SIP Trunking provider

• Phase 2: Connectivity and federation between domains bypassing the PSTN

-Better voice quality

-Its simply cheaper

-More than voice (rich media, presence, messaging)

-Look at: FMC, ENUM & ISN


Pingtel SIPxchange ECS

• SIPxchange is architected around SIP Trunking

• A 2nd Generation system with better technology and superior architecture

• Offering better voice quality

• IT integration without compromise

• Easiest yet to install and manage

• Up to 50% cheaper

The SIP Experts

SIP TrunkingSeminar

Can Your PBX do That?October 2006


Today’s Offerings

• Lack of standards provides for various definitions for “SIP Trunking” - Pingtel is a committed thought leader in the industry involved in defining standards- Recent activity in the SIP Forum Technical Working Group: the “IP PBX and Service

Provider Interoperability Task Group” (http://www.sipforum.org/content/view/179/213) - goal is to specify profiles for the interface between an enterprise PBX and a SIP service provider.

- Pingtel’s Scott Lawrence is leading our involvement in this group - Scott is Pingtel’s lead Consulting Engineer and world-class SIP trunking expert; Scott also serves on the IETF SIP WG and is a project Coordinator at SIPfoundry, the world’s leading open source SIP community; Scott is a leading authority on routing, SIP, security, and network design

• Participation is encouraged by technical contacts at any SIP trunk service provider - effort will dramatically accelerate the adoption of SIP and will be a huge boost to both the PBX and services businesses- Information on SIP Forum membership - www.sipforum.org/index.php?

option=com_content&task=view&id=17&Itemid=47

- Information on “IP PBX and Service Provider Interoperability Task Group” - www.sipforum.org/content/view/179/213)

- In order to contribute to the Technical Working Group you only need individual participant membership, which is free


2 Major SIP Trunk Types

• Type 1: Preferred by Pingtel: Managed IP Connection- SIPxchange work with these offerings today OUT OF

THE BOX (professional services required)

- SIPxchange can use the managed network as primary/backup gateway

- Service provider owns the connection to the customer premise

- Media gateway is in the cloud – aka, “Managed Gateway Services”

- User agent registration not necessary – router needed on premise

- Known offerings currently available and in production

- Level 3, BBCOM, NextGenData, City Voice

- Offerings currently under development

- AT&T, MCI, Time Warner, XO, T Systems

Customer-supplied media gateway

SIPxchange PBX

IPIP

SIP trunk

Max.1,500 seats / server30 media ports

TDM Voice trunk


2 Major SIP Trunk Types (cont’d)

• Not preferred by Pingtel: User Agent Registration- SIPxchange WILL NOT register with

these networks today

- These offers are focused on IP Centrex or Residential applications

- These offerings are not reliable and scaleable enough for SMB market

- User agent registers with provider’s proxy using single user credentials

- Known offerings currently available and in production

- Vonage, Broadvoice, AT&T CallVantage

- Usually used by Asterisk users

Customer-supplied Digium gateway

Asterisk(B2BUA)

IPIP

User Agent registration

Max.100 seats / serverBec. of B2BUA

TDM Voice trunk


Other SIP Trunking Initiatives

• TLS (“transport layer security”; proxy-to-proxy connection)- Encrypts signaling using SSL

- Certificates are used to authenticate connection

- Service Provider equipment and customer prem equipment must both support it

- Pingtel has partially implemented and is looking for partners

• SIPconnect (proxy-to-proxy connection)- All customer premise registrations are sent to service provider

- Service provider can manage all users

- Pingtel is evaluating

- SIP Forum is evaluating before recommending to IETF

- Initiative driven by vendors (e.g., CBeyond, Sylantro)


Summary

• Service providers that authenticate by user registration will not work with SIPxchange today- Pingtel will continue to monitor these providers for offerings targeted at SMB market

• Service providers that support Managed IP services with access control lists can be supported today

- Support of the channel available from Pingtel as Professional Service

• Pingtel is interested in partnering with service providers who wish to support TLS proxy-to-proxy service offerings

• Pingtel will continue to evaluate SIPconnect and work with the SIP Forum and will monitor industry acceptance


Contacts

• Level 3

- Doug Widdoes, Northeast Sales Director, [email protected] , (330) 528-3001

• BBCOM

- Anthony Santana, Account Manager, [email protected]

• NextGenData

- Terry Carpenter, [email protected] , (484) 334-4122

• City Voice

- Kelly Lowther, VP, [email protected] , (603) 659-2912 x104

• Pingtel

- Scott Lawrence, Consulting Engineer, [email protected]

- Neil Segall, VP Sales, [email protected], 781-439-5640

- Bob Pizani, Manager, Systems Engineering, [email protected]

Documents

SIP Trunking Seminar