Time for Telcos to go Beyond POTS!

Transcription

1 Time for Telcos to go Beyond POTS! Live Unified Communication Beyond the Borders See also Live Demo Presentation from ITEXPO SIP Trunking Summit in Miami, February 2011! By: Karl Erik Ståhl President Intertex Data AB CEO and Chairman Ingate Systems AB 2011 Intertex Data AB

2 Where is the Telco Core Business Going? Are Telcos just becoming bandwidth providers? IP is just used to replace POTS Telephony Where is the Live IP Communication: Multimedia or UC? The Beyond POTS islands are taking over: at the Enterprise UC LAN by Skype, Google Talk, Apple and the others 2011 Intertex Data AB 2

3 Where is the Telco Core Business Going? Are Telcos just becoming bandwidth providers? IP is just used to replace POTS Telephony Where is the Live IP Communication: Multimedia or UC? The Beyond POTS islands are taking over: at the Enterprise UC LAN by Skype, Google Talk, Apple and the others Why not better and beyond? Telcos could bring it together and offer better! 2011 Intertex Data AB 3

4 Connect Us Together, add Functionality and Quality! Bringing the Islands together is a Telco core business! So is bringing Functionality, Quality and Reliability! Give us a SIP-addresses (same as ) to each phone number! We cannot get stuck here and not with eight dect phones either! Phones are More! Intertex Data AB 4

5 Quality is Really an Advantage Only the Telcos can Bring! Bringing the Islands together is a Telco core business! so is bringing Functionality, Quality and Reliability! Some basics around IP QoS and why better Internet QoS cannot be for free: a. On the Internet we have Transport layer (4) QoS. The endpoint smartness of TCP makes it all work, filling and sharing the pipe, and backing off for datagram type of packets (e.g. UDP thus RTP). This is mostly often good enough even for voice. However, in the process of sharing a filled pipe, even non TCP packets (e.g. UDP/RTP) are lost (and filling the whole pipe with such packets, is a catastrophe). b. IP Layer (3) QoS (DSCP/TOS bits honored) is available in almost any IP network just ignored on the Internet and gives absolute priority. You simply don t lose any packets unless the whole pipe is filled with your quality level packets (and higher). This is needed for critical real time applications, especially low delay, packet loss sensitive applications; obviously telepresence and sometimes even voice. c. Giving IP Layer (3) QoS to the common Internet for free will of course not help! As soon as the first file sharer will select the highest quality, all users have to do the same to get their share and we are back to a. again. Thus, better IP Layer QoS has to bear a price has to be charged! d. Prioritization and traffic shaping in boxes like ours helps in case a.. However, that only works for traffic that is known or classified by the box, which typically is not the case for SIP using workaround methods like STUN/TURN/ICE or Far End NAT Traversal, Skype, Google Talks or the others and will remain in an environment with the lowest quality. Give us a SIP-address (same as ) for each phone number! We want a real SIP address: sip:john.brown@smartco.com Let us have both: = john.brown@smartco.com And why not the same and SIP address by default with the subscription? 2011 Intertex Data AB 5

6 It can be Done Today It is even Simple! Qwest Deutsche Telecom Internet TeliaSonera Internet AT&T MPLS QoS IP Network QoS IP Network MPLS MPLS SIParator C D R ENUM C D R IX78 Let s see: - What are the problems? - How to do it? 2011 Intertex Data AB 6

7 Why Don t Telcos Offer Global UC Communication? IMS: The thought was good and promised all. But it is complex and so far only used for POTS replication Softswitches: Building PSTN/POTS-like structures on top of IP One major problem: No UC or Multimedia peering between the operators A Voice minute is (maybe) a Voice minute But what is a video minute? Codec, Screen size? Will never happen! And an even worse problem: IMS and SIP do not reach the users on the LANs! Instead FXS ports for analogue phones are still being deployed And SIP trunking of PBXs is hopefully a step in that direction although POTS connectivity is the current level 2011 Intertex Data AB 7

8 There is a Good and Close Solution IP networks are easy to connect globally. Internet is globally connected between Telcos. Higher quality IP networks e.g. for Video Calling can be peered similarly. The CPEs can be clever - like the E-SBCs (Enterprise Session Border Controller) used for SIP Trunking - delivering SIP connectivity into the LAN. Such clever E-SBC CPEs can: Deliver global SIP based UC to the LAN Route the UC communication to other UC endpoints (over IP)! And it can allow access to better IP network than just the Internet And it can produce detailed CDRs for billing! Remember that fixed networks are easy to peer and always will give 10 times more bandwidth at much better quality - at a fraction of the cost - compared to mobile networks. Time to offer high quality global UC Connectivity! (POTS we have had for 100 years.) 2011 Intertex Data AB 8

9 We have Seen Much POTSoIP Gateway US Gateway PBX IP PBX Are we stuck with old POTS telephony over new wires? PSTN Gateway Toll Bypass VPN Tunnel Gateway Soft Switch Very seldom VoIP connectivity between the VoIP IP clouds! Europe IP PBX Voice over Broadband Most broadband VoIP providers still run calls between each other over the PSTN! 2011 Intertex Data AB 9

10 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! 2011 Intertex Data AB 10

11 Or Continue Ahead Until the POTSoIP Peering Collapse Carriers Peer their Networks PSTN Style Conversions back and forth Will increase with time 100 years old Voice quality getting WORSE as more transit peer with VoIP 160 years old Fax service will soon STOP working* - Ticking bomb! * Mike Coffee, CEO of Commetrex: Work in progress by SIP Forum s FoIP Task Group i3 Forum. T.38 works fine in one hop! 2011 Intertex Data AB 11

12 Only IP-Peer via a Common High Quality IP Network! Qwest Comcast Internet TeliaSonera Internet AT&T MPLS *SOFT SWITCH SBC QoS IP Network ENUM *IMS QoS IP Network SBC MPLS MPLS SIParator IX78 * Hoping the SoftSwitch and IMS with their B2BUAs handle more than Voice - full UC 2011 Intertex Data AB 12

13 Or Simply and Better do it at the Edge Even Bill! Qwest Comcast Internet TeliaSonera Internet AT&T MPLS QoS IP Network QoS IP Network MPLS MPLS SIParator C D R ENUM C D R IX Intertex Data AB 13

14 Telcos Often Already have the Key Networks! It is Widely Used for Triple Play Services E.g. Telia E.g. Telia IP-TV Internet IMS IP-TV Internet IMS VoD ADSL PVC2 PVC1 VoIP PVC3 Private Virtual Circuits VoD Ethernet VLAN2 VLAN1 VoIP VLAN3 Virtual LANs (VLAN) E.g. B2 E.g. BT IP-TV VoD Ethernet WAN2 Internet WAN1 IMS VoIP WAN3 IP QoS Separated Subnets IP-TV VoD Priority2 ADSL or Ethernet Internet Priority3 Priority1 IMS VoIP IP Level QoS The Intertex IX78 Supports All of these Architectures! 2011 Intertex Data AB 14

15 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 and other UC! 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 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 2011 Intertex Data AB 15

16 For the Telcos To Do Provide high quality IP pipes for Video and HD Voice (e.g. MPLS) If on separate layer 2 networks for quality, still make them routable to the Internet (for fallback to best effort peered carriers). Enter users in ENUM (public or private) E.164 numbers to SIP address resolution Settlement Free Peering between Carriers for high QoS IP networks Just like for the Internet - Now also for high quality IP network (e.g. by MPLS) Share ENUM (number/sip addresses between the Carriers) Deploy same CPEs (E-SBCs) as for SIP Trunking Can also be general SIP enablers (at least Intertex and Ingate s) for offering all types of SIP based services Process the CDRs from the E-SBC as usual for Billing Via Intertex TR-069 server (ACS) is a very good solution 2011 Intertex Data AB 16

17 Charging for the Better and Net Neutrality Some notes on the CDRs/Taximeter in the E-SBC as implemented in the IX78 a) The taximeter (knowing IP and SIP), monitors and reports the traffic (it does not authenticate or restrict/allow some service usage). b) Classification (set up by the Pipe/Box-Owner or Customer) determines whether the quality pipe shall be used. c) The Pipe/Box-Owner can charge for e.g. Mbps transferred at each quality level and/or for special usage (e.g. video calling) d) Service access is controlled by the Service Provider (e.g. SIP Authentication in INVITE of a video call) e) A Service Provider can get CDRs regarding usage and quality of his service (from the Pipe/Box-Owner) and agree about payment models between Customer, Service Provider and Pipe/Box-Owner (if they want to charge). Services using SIP are classified and monitored already today and can use the quality pipe or the Internet. If another service, like Skype, wishes to offer their Customers guaranteed quality, that service has to show itself so it can be routed through the quality pipe and be monitored. It could simply be based on agreed TOS-bits set by the Skype client, an own Ethernet connector, or a SIP/Skype gateway in the network that is opened by SIP. In the last case the SIP/Skype gateway can simply be given a special SIP outbound proxy, using the quality pipe. The Intertex and Ingate products do such things (in the outbound proxy and in other places) today already:

18 Multiple QoS Separated WAN Pipes Supported (Telia Network) The Multimedia LAN IP-TV VoD Internet IMS VoIP TR-069 All services must be available to multimedia terminals! Over controlled high QoS pipes as well as the Internet. Application Innovation Requires it! VLANs or ADSL Virtual Circuits WiFi The Multimedia LAN Internet IP- PBX PDA Telepresence 18

19 Stop deploying SIP blockers! 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! Internet The 5060 SIP-port is just grabbed on the outside to the FXS ports! Lower level SIP ALGs often cause problems and do not handle more than basic scenarios. Often problems with, or total lack of: SIP to the LAN or WiFi Calls between SIP clients on LAN Calls between internal ATA ports and LAN clients Call transfers, 3-party calls, etc. Using SIP generally over the Internet (Operator took all the SIP ) (Users must not be deprived of general SIP-functionality!) 2011 Intertex Data AB 19

20 Intertex/Ingate CPEs are SIP Capable NAT/Router/Firewalls IMS Internet 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 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.) 2011 Intertex Data AB 20

21 Ingate and Intertex have been SIP Firewall/NAT/ Gateways Picture by since Henry 2002! Sinnreich Here the Worldcom Advantage service /0204voip.html Worldcom 2002a 2011 Intertex Data AB 21

22 Picture by Henry Sinnreich SIP Capable Firewall Ingate and Intertex Integration with existing PBX SIP Trunking of the PBX was part of the solution in 2002 already! 2011 Intertex Data AB 22

23 Intertex IX78 for Volume Deployment In addition to being a router, a firewall, a wireless access point, an ADSL modem etc., the IX78 has several SIP and Telephony related functions: SIP ATA device (2 FXS ports, 1 FXO port) SIP E-SBC Gateway for hosted services LAN and WLAN SIP devices have global SIP connectivity SIP Trunking E-SBC Connecting IP-PBXs directly to operator s SIP Telephony Services Advanced SIP and Telephony routing CDRs for billing. Access to high quality IP networks PBX functionality All these functions can be used together and at the same time! 2011 Intertex Data AB 23

24 And The Ingate Family for Larger Installations Ingate Firewalls and SIParators E-SBC From 50 to simultaneous calls (with media) Used in a wide variety of SIP Trunking installations NAT/Firewall traversal Superior SIP Normalization Multi level security, incl. SIP IDS/IPS QoS (Quality of Service) Failover confgurations 24

25 And Our CPEs Can even be the PBX With full UC! There are many PBXs out there that do not allow Soft Clients, Remote Users or Standard SIP Phones. Registrar PBX with non-sip phones Remote Users Soft Client WiFi Mobile Retire the old PBX PBX

26 And There is Provisioning jkjjk 2011 Intertex Data AB 26

27 2011 Intertex Data AB Element Management System The iems Functions for Provisioning, Monitoring, Reporting, Diagnostics, Logging, Debugging, Support, Configuration and Upgrade. Available now with basic functionality. Will handle both Ingate and Intertex Firewalls and SIParators. Highly scalable, runs on PC servers under the Linux OS. HTTPS/SOAP interface to the IX78. Can read and write all configuration parameters, as well as asynchronous reporting by the device (like SNMP traps). Web based secure access to the iems. Customized portals for operators, installers and customers, for the purpose of administration, management and usage. The iems has northbound interfaces for integrating with the operator s OSS and Fault Management systems, using XML-RPC and/or SOAP.

28 iems CDRs with Call Quality Metrics 2011 Intertex Data AB 28

29 iems CDRs with Call Quality Metrics Now also with Video Call Metrics and Pipe Used! 2011 Intertex Data AB 29

30 3-Party Video Conference with CDRs including Call Quality Metrics 2011 Intertex Data AB 30

31 iems Interfaces OSS, Fault Management, etc. WEB GUI XML-RPC (or SOAP) (GET/SET/EVENTS) CPE CPE CPE Northbound API Southbound API WAN CPE CPE CPE <?xml version="1.0"?> <methodcall> <methodname>settrunk</methodname> <params><param><struct> <member><name>version</name><value>1.0</value></member> <member><name>ems</name><value><struct> <member><name>username</name><value>installer</value> <member><name>password</name><value>foobar123</value></ </struct></value></member> <member><name>service</name><value><struct> <member><name>registrar</name><value>sip.intertex.se</ DB <member><name>proxy</name><value>proxy.intertex.se</value DB DB </struct></value></member> <member><name>trunk</name><value> <array><data> <value><struct> <member><name>identity</name><value> </val <member><name>password</name><value>foobar</value></membe </struct></value> <value><struct> <member><name>identity</name><value> </val <member><name>password</name><value>barfoo</value> </struct></value> </data></array> </value></member> </struct></param></params> CPE </methodcall> 2011 Intertex Data AB 31