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Beyond POTS Replacement Is SIP Trunking a step on that route? © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunking.

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Presentation on theme: "Beyond POTS Replacement Is SIP Trunking a step on that route? © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunking."— Presentation transcript:

1 Beyond POTS Replacement Is SIP Trunking a step on that route? © 2010 Intertex Data AB 1 Prepared for:INTERNET TELEPHONY Conference Ingates SIP Trunking Summit Miami, January 2010 By: Karl Erik Ståhl President & CEO Intertex Data AB Chairman Ingate Systems AB

2 SIP Trunking: Can it be More Than a New Connection? PSTN IP Cloud SIP Trunking Provider IP-PBX Firewall Ingate SIParator® TDM Trunk GW Data & VoIP LAN SIP System SIP Trunk GW

3 and More Than About Interoperability SIP Trunk Ingate SIParator ® -or- Ingate Firewall 3Com Aastra Digium/Asterisk Avaya Cisco Dialogic Ericsson MX-One Fonality Innovaphone Interactive Intelligence Iwatsu Microsoft Mitel NEC / Sphere Nortel Objectworld Panasonic Pingtel Samsung SER Shoretel Siemens SIP-Gear Swyx More in pipeline Networks Airespring AT&T BandTel Bandwidth.com Broadvox Cbeyond Cellip Cordia Corporation Excel Switching Gamma Global Crossing IP-Only Juma Networks Level 3 Netlogic Nexvortex Nuvox O1 Paetec Primus RNK Telecom TDC Tele2 Telia Toplink VoEX VoIP Unlimited Voxbone More in pipeline..... Carrier Equipment Acme Packet Broadsoft NexPoint Sonus Sylantro SER Compliant with Service providersIP-PBXs See:

4 © 2010 Intertex Data AB 4 Installation Wizard and More Than Easy Deployment Update

5 © 2010 Intertex Data AB 5 Benefits of SIP Trunking Monthly cost savings Single network for all communications Lower cost of Moves, Adds and Changes Disaster Recovery / Business Continuity User provisioning Steps of going beyond POTS replacement – Unified Communication Mobility – Remote workers Multimedia - Video, IM, Presence, Real Time Text RFC 4103, etc. Real SIP address – like address WiFi mobile phone communication Lets talk about this now!

6 © 2010 Intertex Data AB 6 There is Potential to Go Beyond! RJ45 LAN Intranet Internet Now we have a new 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 And we have a new standard: SIP And there is more than Voice: Presence, IM, Video, etc.

7 © 2010 Intertex Data AB 7 Europe US VPN Tunnel IP PBX PBX But have We Seen Much More than 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?

8 © 2010 Intertex Data AB 8 HTTP created the Web SMTP created SIP can create global Live IP Person-to-Person Communication! And When will We See the Next Step of Internet Usage?

9 © 2010 Intertex Data AB 9 There is a Severe Infrastructure Problem… LAN FW Internet web SIP does not traverse the common NATs and firewalls protecting the LANs. IMS (SIP based) IMS (SIP based) What about SIP for Live Person-to-Person Communication? A common Network and common Protocols changed our lives: SMTP gave us global ! HTTP gave us the Web! NATs and Firewalls were designed to allow such protocols.

10 © 2010 Intertex Data AB 10 Why are NATs and Firewalls Such Obstacles Typical Internet protocol (SMTP, HTTP…) Internet HOST SERVER SIP (and H.323…) connects Person-to-Person Internet PERSON SIP is the Protocol for IP Communication Person-to-Person, BUT IT DOES NOT REACH THE USERs! Locate the personSet up a session + Open real time media streams +

11 Data & VoIP LAN Soft Clients and Multimedia Terminals PSTN Public Internet SIP Trunking Provider GW IP-PBX Firewall SIP Trunking does not pass a SIP unaware NAT/firewall! …and the firewall cannot be opened enough to make it work because of NAT. SIP System And that is a Main Problem when SIP Trunking IP-PBXs

12 © 2010 Intertex Data AB 12 And Hosted VoIP Suffers from the Same Problem 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!

13 © 2010 Intertex Data AB 13 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. Lets Use Real SIP Capable NAT/Router/Firewalls Internet Problems solved where they occur No special requirements on the SIP Client – Just standard SIP Wired or wireless SIP clients (phones, soft clients, PDAs) SIP Intertex and Ingate CPEs have a SIP Proxy based Firewall/NAT General, can handle complex call scenarios and all SIP services Additional functionality available (PBX like functionality, ENUM, etc.) IMS

14 © 2010 Intertex Data AB 14 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 Lets Fix the SIP Trunking and at the Same Time Enable Going Beyond POTS Replication

15 © 2010 Intertex Data AB 15 And Step in to the 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

16 © 2010 Intertex Data AB 16 Internet THIS LAN, SIP Trunking Summit US, Miami Multimedia Voice Video Real-time Text RFC4103 Omnitor Case Study: Beyond POTS: Mobility, Multimedia and Numbers Sweden ADSL

17 Gunnar Hellström, Omnitor, Presenting Live from Sweden Using Omnitor application Allan eC: Voice: G.722 wide band codec Video: H kbps Real-time text: RFC4103 Using standard SIP over the Internet. See presentation: Omnitor- TotalConversation Other Live Demos Follow!

18 INGATE LAN ingate.com Internet US, Miami THIS LAN, SIP Trunking 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 Sweden ADSL

19 © 2010 Intertex Data AB 19 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

20 INGATE LAN ingate.com Internet US, Miami THIS LAN, SIP Trunking 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 Sweden ADSL

21 © 2010 Intertex Data AB 21 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

22 INGATE LAN ingate.com Internet US, Miami THIS LAN, SIP Trunking 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 to the local switch here) and we saw global SIP calls – not restricted to own domain Sweden ADSL

23 © 2010 Intertex Data AB 23 Beyond POTS: Mobility, Multimedia and Numbers Telephone numbers WILL be around for long -We are simple 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!

24 INGATE LAN ingate.com Internet US, Miami THIS LAN, SIP Trunking 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) Sweden ADSL

25 © 2010 Intertex Data AB 25 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:

26 © 2010 Intertex Data AB 26 SIP Capable Firewalls Ingate Systems Inc. 7 Farley Road Hollis NH United States Ph: +1 (603) Ph Sweden: Intertex Data AB Rissneleden 45 SE Sundbyberg Sweden Tel: See us at ITEXPO Room A108!

27 © 2010 Intertex Data AB 27 STUN, TURN, ICE (client based) and 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, you need something reliable and secure that also handles real complex call scenarios What about STUN, TURN, ICE and Far End Nat Traversal (FENT)?

28 © 2010 Intertex Data AB 28 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. FW is no longer in control of what is allowed into the LAN! STUN, TURN and ICE delegate control to the Client and can also be used for evil protocols. 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). Both 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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