Live Unified Communication Beyond the Borders UC Across the Borders Live Unified Communication Beyond the Borders Prepared for: INTERNET TELEPHONY Conference Ingate’s SIP Trunk-UC Summit Miami, Februari 2011 By: Karl Erik Ståhl President Intertex Data AB Chairman Ingate Systems AB karl.stahl@intertex.se © 2010 Intertex Data AB 1

by in the US Intertex & Ingate Same parent company Intertex: SMB, SOHO and home SIP Firewalls and E-SBCs For volume deployment Ingate: Enterprise and SMB SIP Firewalls and E-SBCs SIParators® for enterprises and projects Cooperation in management and development Co-developed SIP code Ingate represents Intertex in the US by in the US

SIP Trunking – Now SIP Trunk-UC Summit UC, Unified Communication – Many definitions… This session is about the Live (Real Time) Person-to-Person part (other parts may be Web and Email based) Telephony – VoIP, SIP Trunking Video, HD voice Presence IM – Instant Messaging Today’s SIP Trunking makes VoIP global, but it is still mostly POTS (Plain Old Telephony Service) But for the better; Video, better Voice, Presence and IM, we mostly see local islands of UC

We have Seen Much POTSoIP Gateway Toll Bypass Europe US VPN Tunnel IP PBX IP PBX Gateway PBX Are we stuck with old POTS telephony over new wires? PSTN Gateway Voice over Broadband Soft Switch Very seldom VoIP connectivity between the VoIP IP clouds! Most broadband VoIP providers still run calls between each other over the PSTN!

we are building UC islands inside our firewalls IP Cloud And in the enterprise we are building UC islands inside our firewalls Firewall PSTN The old PSTN has brought the islands together

brings the VoIP together PSTN SIP Trunk Provider 1 SIP Trunk Provider 2 IP Cloud Firewall PSTN SIP TRUNKING brings the VoIP together For VIDEO, IM, PRECENSE etc., there is no “PSTN” to connect with. Do it IP style! – Simply use SIP!

UC Shall Not Reside in LAN Islands Only! Global IP Connectivity VoIP++ All SIP Services PSTN-style structures are good for talking to the PSTN world, not for UC. Using SIP in its IP-style connects globally and has lots of applications. It’s not magic – It’s just the SIP standard!

Let’s put it to use! Demos will follow Back to Basics The IP networks (Internet and other) are connected There is a standard, SIP SIP (incl. SIMPLE) is general, for Live Person-to-Person communication, POTS replication is just one usage But SIP communication must reach the users on the protected LANs behind NAT/Firewalls! Some E-SBCs can provide general SIP traversal of NATs and Firewalls And some are general session routing SIP Proxies Let’s put it to use! Demos will follow

Our E-SBCs are for More than SIP Trunking! IMS Internet SIP No battery draining of WiFi mobile phones, otherwise caused by keep-alive packets* inhibiting sleep mode. * Work-around methods for SIP NAT-traversal like STUN, TURN, ICE and Far End NAT Traversal use frequent keep-alive packets to keep holes in the NAT/Firewall open. Problems solved where they occur Wired or wireless SIP clients (phones, soft clients, PDAs) No special requirements on the SIP Client – Just standard SIP Intertex and Ingate have SIP Proxy based SIP aware Firewall/NATs General, can handle complex call scenarios and all SIP services Routes calls – as SIP proxies shall do (not just passing all to an ITSP) Additional functionality available (SIP server, PBX functionality etc.) 9

And it is More than Multimedia! Mobility We certainly want our home workers connected to the company PBX And the same goes for our road warriors All should have all PBX services Multimedia Adding Video, IM and Presence And HD Voice And Telepresence Numbers Sure we need the old telephone numbers But also give us usable SIP addresses – Just like email addresses!

Internet Japan US, Miami Sweden kamill@von.sipnr.org Internet PSTN SIP/PSTN Gateway SIP Trunk Provider 2 US, Miami INTERTEX LAN intertex.se Sweden THIS LAN, SIP Trunk-UC Summit calle@intertex.se sophie@ingate.com steeg@intertex.se PSTN SIP/PSTN Gateway SIP Trunk Provider 1 3G stefan@ingate.com INGATE LAN ingate.com CELL PSTN Let’s see what we can show on an Internet connection here and now at IT EXPO in Miami!

Beyond POTS: Mobility, Multimedia and Numbers We certainly want our home workers connected to the company PBX And the same goes for our road warriors at the hotel at public WiFi All should have all PBX services Reached by extension number or DID Place PSTN calls (displaying correct CallerID) Voice mail, conferencing etc. Presence, IM, video if supported by the PBX

Internet Japan US, Miami Sweden PSTN PSTN SIP/PSTN Gateway SIP/PSTN kamill@von.sipnr.org Internet PSTN SIP/PSTN Gateway SIP Trunk Provider 2 US, Miami INTERTEX LAN intertex.se Sweden PSTN SIP/PSTN Gateway SIP Trunk Provider 1 3G stefan@ingate.com INGATE LAN ingate.com CELL PSTN THIS LAN, SIP Trunk-UC Summit (sophie@ingate.com) steeg@intertex.se calle@intertex.se PBX Mobility with SIP Trunking (demo) PSTN  +46 8 12345629 my direct number steeg  29 = my extension number calle  23 (steeg) PSTN  +46 8 12345600 Intertex main  ext 29, 25s leave Voice Mail calle  mobile in the hall Voice Mail comes via email

Beyond POTS: Mobility, Multimedia and Numbers Presence is really useful So is IM (Instant Messaging) Laptops have cameras and good screens, so why not video? Video conferencing does not have to be complex with huge cost and for internal use only. And voice can actually be better than 3kHz AM-radio quality! Who said MOS score 5 was perfect? Hardly HiFi?

Internet Japan US, Miami Sweden PSTN PSTN SIP/PSTN Gateway SIP/PSTN kamill@von.sipnr.org Internet PSTN SIP/PSTN Gateway SIP Trunk Provider 2 US, Miami INTERTEX LAN intertex.se Sweden PSTN SIP/PSTN Gateway SIP Trunk Provider 1 3G stefan@ingate.com INGATE LAN ingate.com CELL PSTN THIS LAN, SIP Trunk-UC Summit sophie@ingate.com (steeg@intertex.se) calle@intertex.se …and other SIP based applications (demo) Presence, Instant Messaging ((is kamill or other on-line?) Not restricted to own domain intertex.se, here also ingate.com calle@intertex.se  sophie@ingate.com (listen + video) Wide band codec: “S” is not “F” anymore! Video Media goes the shortest way (just trough the local switch here) and we saw global SIP calls – not restricted to own domain

Beyond POTS: Mobility, Multimedia and Numbers Telephone numbers WILL be around for a long time We are simply too used to E.164 numbers and everyone has one But they are really not particularly user friendly… Would email have been a success if we had used our fax numbers? Operators often provide SIP names like u081234567@sip14.provider.se Not user friendly at all. For internal use only. We want a real SIP address: john.brown@smartco.com Just like our email addresses Let us have both: +46 8 1234567 = john.brown@smartco.com! Service providers can do it Here the Intertex and Ingate products do it!

Internet Japan US, Miami Sweden PSTN PSTN SIP/PSTN Gateway SIP/PSTN kamill@von.sipnr.org Internet PSTN SIP/PSTN Gateway SIP Trunk Provider 2 US, Miami INTERTEX LAN intertex.se Sweden PSTN SIP/PSTN Gateway SIP Trunk Provider 1 3G stefan@ingate.com INGATE LAN ingate.com CELL PSTN THIS LAN, SIP Trunk-UC Summit 0850004123@ipkund.rixtelecom.se sophie@ingate.com calle@intertex.se Telephone numbers and SIP addresses (demo) Can we do global SIP calls over the SIP trunk? It is up to the operators! E.g. Telia routes real SIP calls and don’t steal the media (even though they are on a managed VoIP cloud) 0850004123  calle using 08 12345629 (IP  PSTN ------> PSTN  IP  only POTS voice) sophie  calle using 08 12345629 (ENUM: IP  IP  quick, wide band codec, video)

ENUM – Using Phone Numbers but Staying on IP 2) ENUM lookup: Is there a SIP address for +46812345629? Ask DNS: 9.2.6.5.4.3.2.1.8.6.4.e164.arpa Yeah try sip:calle@intertex.se 3) Place the call directly to: sip:calle@intertex.se IP IP 1) Dial Phone Number 08 12345629 +46 8 12345629 calle@intertex.se PSTN Not only for PSTN by-pass, but also for better voice and multimedia Clients, Intertexes/Ingates, or service providers can use ENUM

Telcos More Than Bandwidth Providers? Telco Opportunity Video Calling High Quality, Chargeable, Global Video Calling Ready to go, using SIP Trunking Infrastructure High Quality (Telepresence) Video Calling Routed and Billed (CDRs produced) by the E-SBC Simple settlement free IP Peering between Telcos

What’s Special About Video Calling? We have been building islands – again… But there is no old Video PSTN to connect those together However, there is a standard (SIP) and a network (Internet) We have seen such video calls for a long time What more is needed? High quality – Teleprecense; Guaranteed bandwidth and QoS? Global; Not only within a company and not only within one carrier’s network Telephone numbers (in addition to sip addresses) Allow Telcos to Bill (being more than just Bandwidth Providers)? 20

There is a Solution! 21 Do More at the Enterprise Edge! We can route here – The earlier the better We can produce CDR’s for billing here We can do number resolution here (or the ITSP can do it) The Good News: Reuse the SIP Trunking infrastructure (using E-SBCs) Simple peering between carriers 21

Reusing the SIP Trunking E-SBC Telco owned E-SBCs are already used for (voice) SIP Trunking Full operator control Service provider’s demarcation point Enables the SIP Trunking – Video is not different from voice for: NAT/Firewall traversal, PBX interoperability and Security Reuse the same E-SBC for Video Calling! In the Ingate and Intertex E-SBCs, it is all there: Classify outgoing calls (as Video, HD voice or plain voice) Assure right quality pipe and/or quality marking is used Route the call directly to the other party (or Use ENUM (public or private) for E.164 number to SIP address resolution Only settlement free IP peering between operators required Can fallback to best effort IP peering (Internet) in operator network Produce and deliver CDRs for each call Report Minutes and Data used Include video and voice quality metrics (including MOS scores) Deliver via Radius, Syslog, Management system (TR-069 informs) or method by choice 22

Simple For the Carrier Qwest Internet AT&T Internet SIParator IX78 MPLS QoS IP Network QoS IP Network MPLS ENUM CDR CDR SIParator IX78

The Intertex IX78 Supports All of these Architectures! Quality Separated Networks Out to the Customer Edge is Nothing New It is Widely Used for Triple Play Services Private Virtual Circuits E.g. Telia Internet ADSL PVC1 IP-TV VoD IMS VoIP PVC2 PVC3 E.g. Telia Internet Ethernet VLAN1 IP-TV VoD IMS VoIP VLAN2 VLAN3 Virtual LANs (VLAN) E.g. B2 Internet Ethernet WAN1 IP-TV VoD IMS VoIP WAN2 WAN3 IP QoS Separated Subnets IP Level QoS E.g. BT Internet ADSL or Ethernet Priority3 Priority2 Priority1 IMS VoIP IP-TV VoD The Intertex IX78 Supports All of these Architectures! 24

Telcos Providing More than Bandwidth? Operators deploy CPEs (E-SBCs) for SIP Trunking Can also be general SIP enablers (at least Intertex’s and Ingate’s) Provide high quality pipes for live communication! If on separate layer 2 networks for quality, still make them routable to the Internet. Provide Presence Server! Per-to-peer presence is not good enough (heavy signaling, difficulties maintaining sync.) Allow customers to manage their buddy lists and call policies Provide the SIP Server and more if you wish SIP Services can be anywhere (with cured firewall problem)! Our E-SBCs produces CDRs if the provider wishes to bill The CDRs also include bytes transferred & Call Metrics (e.g. MOS) 25

SIP Capable Firewalls See us at ITEXPO Room 208A! Intertex Data AB www.intertex.se info@intertex.se Rissneleden 45 SE-174 44 Sundbyberg Sweden sip:reception@intertex.se Tel: +46 8 6282828 Ingate Systems Inc. www.ingate.com Info@ingate.com 7 Farley Road Hollis, NH 03049 United States Ph: +1 (603) 883-6569 Tel sv: +46 8 6007750