1 / 16

SIP Trunking Seminar

SIP Trunking Seminar. Can Your PBX do That? October 2006. 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?

simon-mcintosh
Download Presentation

SIP Trunking Seminar

An Image/Link below is provided (as is) to download presentation Download Policy: Content on the Website is provided to you AS IS for your information and personal use and may not be sold / licensed / shared on other websites without getting consent from its author. Content is provided to you AS IS for your information and personal use only. Download presentation by click this link. While downloading, if for some reason you are not able to download a presentation, the publisher may have deleted the file from their server. During download, if you can't get a presentation, the file might be deleted by the publisher.

E N D

Presentation Transcript

  1. SIP TrunkingSeminar Can Your PBX do That? October 2006

  2. 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

  3. 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

  4. SIP SIP SIP SIP SIP SIP Lines vs. Trunks Poor Man’s Trunking – Good for small deployments • No DIDs • Security through registration (like phones) • Holes in your firewall • STUN keep-alive traffic Holes 508-345-1123 B2BUA 508-345-6532 508-345-9987 ITSP

  5. Load Balance & Fail Over Deployment Example sipX HA Call Control Server Ingate SIParator sipX Media & Config Server • 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

  6. 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

  7. 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

  8. 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

  9. 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

  10. SIP TrunkingSeminar Can Your PBX do That? October 2006

  11. 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

  12. 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 SIPxchange PBX SIP trunk TDM Voice trunk Customer-supplied media gateway IP Max. 1,500 seats / server 30 media ports

  13. 2 Major SIP Trunk Types (cont’d) User Agent registration • 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 Asterisk (B2BUA) TDM Voice trunk Customer-supplied Digium gateway IP Max. 100 seats / server Bec. of B2BUA

  14. 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)

  15. 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

  16. Contacts • Level 3 • Doug Widdoes, Northeast Sales Director, doug.widdoes@level3.com , (330) 528-3001 • BBCOM • Anthony Santana, Account Manager, asantana@bbcominc.com • NextGenData • Terry Carpenter, tlc2@nextgendata.com , (484) 334-4122 • City Voice • Kelly Lowther, VP, kelly@starshipcorp.com , (603) 659-2912 x104 • Pingtel • Scott Lawrence, Consulting Engineer, slawrence@pingtel.com • Neil Segall, VP Sales, nsegall@pingtel.com, 781-439-5640 • Bob Pizani, Manager, Systems Engineering, bpizani@pingtel.com

More Related