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Enterprise-Centric UC Live Unified Communication Beyond the Borders © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunk-UC.

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Presentation on theme: "Enterprise-Centric UC Live Unified Communication Beyond the Borders © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunk-UC."— Presentation transcript:

1 Enterprise-Centric UC Live Unified Communication Beyond the Borders © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunk-UC Summit Los Angeles, October 2010 By: Karl Erik Ståhl President Intertex Data AB Chairman Ingate Systems AB

2 © 2010 Intertex Data AB 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 2 by in the US

3 © 2010 Intertex Data AB 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 based) Telephony – VoIP, SIP Trunking Video, HD voice Presence IM – Instant Messaging Todays 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 3

4 © 2010 Intertex Data AB Some History (Before the Internet) MHS, Message Handling Systems appeared where terminals or computers where connected One started building gateways between offices and partners (Compare todays Federation) Standard required! Telcos came up with X.400 Store and forward messages between Telcos, via various networks Extensive OSI layered standard – Complex! Chargeable (good for the Telcos, they thought) Then came the Internet with its simple SMTP for One network & standard, global connectivity (no islands) The revolution (explosion) X.400 and proprietary MHS died 4

5 © 2010 Intertex Data AB The Web and Further The World Wide Web, with its HTTP standard, created something totally new that we today cannot be without Killed off the Videotex services and Frances successful Minitel World Wide = global No island! Neither nor the Web are chargeable in themselves Telcos became bandwidth providers… What was next to come on the Internet? Live (Real Time) communication between persons! H.323 came with Video Telephony H.323 was much like X.400 – Not internet style SIP is the Internet protocol! 5

6 © 2010 Intertex Data AB 6 HTTP created the Web SMTP created SIP should create global Live IP Person-to-Person Communication! The Next Step of Internet Usage

7 © 2010 Intertex Data AB 7 …but NATs and Firewalls are an Infrastructure Problem SIP (and H.323…) connects Person-to-Person Internet PERSON Locate the personSet up a session + Open real time media streams + Typical Internet protocol (SMTP, HTTP…) Internet HOST SERVER NAT/Firewall SIP is the Protocol for IP Communication Person-to-Person, BUT IT DOES NOT REACH THE USERS!

8 © 2010 Intertex Data AB So What Happened? While there has been great success for MSN, Skype and local enterprise live UC (using proprietary protocols)… Telcos have used SIP to replicate POTS (POTSoIP) Got stuck in replacing parts of the PSTN Islands again Telcos cant even give their broadband customers a proper SIP address like Are we leaving it all to Skype (very good at penetrating firewalls)? 8 Go better and beyond!

9 © 2010 Intertex Data AB 9 Europe US VPN Tunnel IP PBX PBX We have Seen Much POTSoIP PSTN Gateway Toll Bypass IP PBX Gateway Soft Switch Gateway Voice over Broadband Very seldom VoIP connectivity between the VoIP IP clouds! Most broadband VoIP providers still run calls between each other over the PSTN! Are we stuck with old POTS telephony over new wires?

10 © 2010 Intertex Data AB 10 Telcos Roll out CPEs where SIP Ends Up in Old Phones Internet The 5060 SIP-port is just grabbed on the outside to the FXS ports! (And lower level SIP ALGs often cause problems and do not handle more than basic scenarios.) Telephone ports (FXS) on the CPE is a popular way to deploy IP telephony. By logically placing the SIP clients on the outside of the NAT/Firewall, unreliable work-around methods like STUN, TURN and ICE become unnecessary. However, this only gives POTS replication, often even stopping general SIP based services! FXS ports (for plugging in analog phones) is really POTS replication!

11 © 2010 Intertex Data AB 11 We Want a World of Global Live IP Communication Fix the NATs and firewalls and there is no reason to be caught in POTSoIPs islands! SIP connects globally and has lots of applications. Its not magic – Its just the SIP standard! VoIP++ Global IP Connectivity All SIP Services

12 © 2010 Intertex Data AB 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 it must reach the users on the protected LANs behind NAT/Firewalls! Some E-SBCs can provide general SIP traversal NATs and Firewalls The Intertex and Ingate products do that, in addition to the SIP trunking (you dont have to choose only one) Lets put it to use!Demos will follow 12

13 © 2010 Intertex Data AB Is it about SIP Trunking, Hosted Services or a Combination? The Trunk Service is in the Cloud, while the PBX service (as the users see it) is on the LAN. That is already a combination, that SIP Trunking – for Telephony - brought together on a Global level. The other Live parts of UC; Video, better Voice, Presence, IM, also need to be brought together on a Global level - Not having it locked into enterprise islands! Todays demonstrations will show that it can be done by following the SIP standard and using the E-SBCs at the enterprise edge, to allow UC SIP communication across the borders (the enterprise firewalls).

14 © 2010 Intertex Data AB 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. Our CPEs are SIP Capable NAT/Router/Firewalls Internet Problems solved where they occur Wired or wireless SIP clients (phones, soft clients, PDAs) No special requirements on the SIP Client – Just standard SIP SIP Intertex and Ingate have SIP Proxy based SIP aware Firewall/NATs General, can handle complex call scenarios and all SIP services Additional functionality available (SIP server, PBX functionality etc.) IMS

15 © 2010 Intertex Data AB And the CPEs are also Adapted for SIP Trunking PSTN Public Internet SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX Demarcation point of service and bringing SIP communication to the LAN Soft Clients and Multimedia Terminals Intertex IX78 Remote Users

16 © 2010 Intertex Data AB For SIP Trunking, the Service is in the Cloud PSTN SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX Service in the Cloud Users on the LAN

17 © 2010 Intertex Data AB For (Remote) Users, the Service is on the LAN PSTN SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX Remote Users …and users on the LAN Service on the LAN User in the Cloud

18 © 2010 Intertex Data AB And Just Some Part of the UC Service may be in the Cloud PSTN SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX Service on the LAN UC Voice Mail One example is MS Exchange UM for the BPOS service: Voice Mails are recorded and played using SIP with TLS and SRTP. Specific Service in the Cloud, e.g. Voice Mail, Presence server, etc.

19 © 2010 Intertex Data AB SIP Must Work with Services and Users Everywhere! PSTN SIP Trunking Provider GW SIP System Data & VoIP LAN IP-PBX UC Voice Mail Remote Users SIParator® Firewall Ingate/Intertex E-SBCs enable SIP based Live UC Across the Borders! (SIP does not traverse ordinary NAT/Firewalls.)

20 © 2010 Intertex Data AB 20 Can We Move Beyond POTS Today? RJ45 LAN Intranet Internet We have a global network: The IP Networks RJ11 POTS and PSTN have been there for 100 years Black Phone IP Phone 3.5 kHz isnt HiFi, but MOS is 5! Soft Client WiFi Mobile We have a standard: SIP And there is more than Voice: Presence, IM, Video, etc.

21 INGATE LAN ingate.com Internet US, Los Angeles THIS LAN, SIP Trunk-UC Summit CELL PSTN INTERTEX LAN intertex.se Sweden 3G PSTN SIP/PSTN Gateway SIP Trunk Provider 1 PSTN SIP/PSTN Gateway SIP Trunk Provider 2 Japan

22 © 2010 Intertex Data AB 22 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

23 INGATE LAN ingate.com Internet US, Los Angeles THIS LAN, SIP Trunk-UC Summit CELL PSTN INTERTEX LAN intertex.se Sweden 3G PSTN SIP/PSTN Gateway SIP Trunk Provider 1 PSTN SIP/PSTN Gateway SIP Trunk Provider 2 PBX Mobility with SIP Trunking (demo) PSTN my direct number steeg 29 = my extension number calle 23 (steeg) PSTN Intertex main ext 29, 25s leave Voice Mail Calle mobile in the hall Voice Mail comes via Japan

24 © 2010 Intertex Data AB 24 Beyond POTS: Mobility, Multimedia and Numbers 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? Presence is really useful

25 INGATE LAN ingate.com Internet US, Los Angeles THIS LAN, SIP Trunk-UC Summit CELL PSTN INTERTEX LAN intertex.se Sweden 3G PSTN SIP/PSTN Gateway SIP Trunk Provider 1 PSTN SIP/PSTN Gateway SIP Trunk Provider 2 …and other SIP based applications (demo) Presence, Instant Messaging (Who is available?) Not restricted to own domain intertex.se, here also 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 Japan

26 © 2010 Intertex Data AB 26 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 have been a success if we had used our fax numbers? Operators often provide SIP names like -Not user friendly at all. For internal use only. We want a real SIP address: -Just like our addresses Let us have both: = -Service providers can do it -Here the Intertex and Ingate products do it!

27 INGATE LAN ingate.com Internet US, Los Angeles THIS LAN, SIP Trunk-UC Summit CELL PSTN INTERTEX LAN intertex.se Sweden 3G PSTN SIP/PSTN Gateway SIP Trunk Provider 1 PSTN SIP/PSTN Gateway SIP Trunk Provider 2 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 dont steal the media (even though they are on a managed VoIP cloud) Calle using (IP PSTN > PSTN IP only POTS voice) sophie Calle using (ENUM: IP IP quick, wide band codec, video) Japan

28 © 2010 Intertex Data AB 28 IP PSTN ENUM – Using Phone Numbers but Staying on IP IP Not only for PSTN by-pass, but also for better voice and multimedia Clients, Intertexes/Ingates, or service providers can use ENUM ) ENUM lookup: Is there a SIP address for ? Ask DNS: e164.arpa Yeah try 1) Dial Phone Number ) Place the call directly to:

29 © 2010 Intertex Data AB Telcos Providing More than Bandwidth? Operators deploy CPEs (E-SBCs) for SIP Trunking Can also be general SIP enablers (at least Intertexs and Ingates) 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) 29

30 © 2010 Intertex Data AB 30 SIP Capable Firewalls Ingate Systems Inc. 7 Farley Road Hollis, NH United States Ph: +1 (603) Tel sv: Intertex Data AB Rissneleden 45 SE Sundbyberg Sweden Tel: See us at ITEXPO Room 403A!

31 © 2010 Intertex Data AB 31 STUN, TURN, ICE (client based) and Far End Nat Traversal (FENT) (typically done by SBCs) are alternative methods for working around non SIP capable NATs and Firewalls Use them if required, e.g. for road warriors behind well behaved NATs with a not too tight firewalls Ingate and Intertex can enable FENT to help SIP remote clients behind non SIP aware NATs and firewalls, e.g. Remote Users But for SIP trunking and global and general SIP communication, one needs something reliable and secure that also handles real complex call scenarios What about STUN, TURN, ICE and Far End Nat Traversal (FENT)?

32 © 2010 Intertex Data AB 32 Workaround Methods have their Limitations… IMS VoIP IMS LAN FW RELIABILITY: STUN, TURN, ICE and Far End NAT Traversal (FENT) rely on guesswork of NAT/Firewall behavior – Thus never fully reliable. Unsuccessful calls – especially in complex scenarios, one way media, timeout during calls etc. etc.. Internet Keep-alive packets inhibit sleep mode, thus draining batteries of WiFi devices. STUN TURN SECURITY POLICY: These workarounds require Firewalls to have large port ranges open from inside. Enterprises can therefore not maintain tight firewalls and have same strict control! STUN, TURN and ICE delegate control to the Client. FENT delegates control to the Operator. No control of QoS– where it is most important! SECURITY AND STABILITY: STUN, TURN, ICE are Client based, FENT is operator based (part of SBC). All rely on punching holes in the Firewall and keeping NAT bindings open. Issues: And with general SIP on several WAN-pipes: No chance!


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