time for telcos to go beyond pots! - uc...

Time for Telcos to go Beyond POTS!Live Unified Communication Beyond the Borders

© 2011 Intertex Data AB

See also Live Demo Presentation from ITEXPO SIP Trunking Summit in Miami, February 2011!

By: Karl Erik Ståhl President Intertex Data ABCEO and Chairman Ingate Systems [email protected]

© 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 TelephonyWhere 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


Where is the Telco Core Business Going?

Telcos could bring it together and offer better!

Why not better and beyond?

Are Telcos just becoming bandwidth providers?IP is just used to replace POTS TelephonyWhere 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


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 email) to each phone number!

and not with eight dect phones either!

We cannot get stuck here

http://www.avaya.com/usa/campaign/avaya-flare-experience-guided-tour/

Phones are More!


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; obviouslytelepresence 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 email) for each phone number!We want a real SIP address: sip:[email protected] us have both: +46 8 1234567 = [email protected] why not the same email and SIP address by default with the subscription?


It can be Done Today – It is even Simple!

TeliaSonera Internet

SIParator IX78

ENUM

QoS IP Network QoS IP Network

CDR

CDR

MPLSMPLS

AT&TQwest Deutsche Telecom Internet

MPLS

Let’s see:- What are the problems?- How to do it?


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 minuteBut 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 deployedAnd SIP trunking of PBXs is hopefully a step in that direction – although POTS connectivity is the current level


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 LANRoute the UC communication to other UC endpoints (over IP)!And it can allow access to better IP network than just the InternetAnd 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.)


Europe

US

VPNTunnel

IP PBX

PBX

We have Seen Much POTSoIP

PSTN

Gateway

Gateway

TollBypass

IP PBX

Gateway

SoftSwitch

Gateway

Voice overBroadband

Very seldom VoIP connectivity between the VoIP IP clouds!

Most broadband VoIP providers still run calls between each other over the PSTN!

Are we stuckwith old POTStelephony over new wires?


UC Shall Not Reside in LAN Islands Only!

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!

VoIP++

Global IP Connectivity

All SIP Services


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 VoIP160 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!


Only IP-Peer via a Common High Quality IP Network!


SIParator IX78

ENUM

QoS IP Network QoS IP Network MPLSMPLS

AT&TQwestComcast Internet

MPLS

SBCSBC

*SOFT SWITCH

*IMS

* Hoping the SoftSwitch and IMS with their B2BUAs handle more than Voice - full UC


Or Simply and Better do it at the Edge – Even Bill!


SIParator IX78

ENUM

QoS IP Network QoS IP Network

CDR

CDR

MPLSMPLS

AT&TQwest

MPLS

Comcast Internet


Telcos Often Already have the Key Networks!It is Widely Used for Triple Play Services

The Intertex IX78 Supports All of these Architectures!

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

Priority3Priority2 Priority1

IMSVoIP

IP-TVVoD


Reusing the SIP Trunking E-SBC

Telco owned E-SBCs are already used for (voice) SIP TrunkingFull operator controlService provider’s demarcation pointEnables 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 usedRoute 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


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 networksJust 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 TrunkingCan 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 BillingVia Intertex’ TR-069 server (ACS) is a very good solution

Charging for the Better and Net NeutralitySome 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)

TR-069Internet

IP-TVVoD

IMSVoIP

PDA

All services must be available to multimedia terminals! – Over controlled high QoS pipes as well as the Internet.

VLANs or ADSL Virtual Circuits

The Multimedia LAN

The Multimedia LAN

WiFi

Internet

Application Innovation Requires it!

Telepresence

IP-PBX


Stop deploying SIP blockers!

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.

• 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!)

Often problems with, or total lack of:

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!


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.

Intertex/Ingate CPEs are SIP Capable NAT/Router/Firewalls

Internet

Problems solved where they occurWired 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/NATsGeneral, can handle complex call scenarios and all SIP servicesAdditional functionality available (SIP server, PBX functionality etc.)

IMS


Worldcom 2002a

Picture by Henry Sinnreich

Ingate and Intertex have been SIP Firewall/NAT/ Gateways since 2002!Here the Worldcom ”Advantage” servicehttp://www.nwfusion.com/news/2002/0204voip.html


Picture by Henry Sinnreich

Integration with existing PBX

SIP Capable FirewallIngate and Intertex

SIP Trunking of the PBX was part of the solution in 2002 already!


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!

24

And The Ingate Family for Larger Installations

Ingate Firewalls and SIParators® – E-SBC From 50 to 3 000 simultaneous calls (with media)Used in a wide variety of SIP Trunking installationsNAT/Firewall traversalSuperior SIP NormalizationMulti level security, incl. SIP IDS/IPSQoS (Quality of Service)Failover confgurations

PBX with non-SIP phones

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

Soft Client WiFi Mobile

Remote Users

PBXRetire the old PBX…


And There is Provisioning… jkjjk


Element Management System – The iEMSFunctions 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.


iEMS – CDRs with Call Quality Metrics


iEMS – CDRs with Call Quality MetricsNow also with Video Call Metrics and Pipe Used!


3-Party Video Conference with CDRs including Call Quality Metrics


iEMS Interfaces<?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</

<member><name>proxy</name><value>proxy.intertex.se</value </struct></value></member><member><name>trunk</name><value><array><data>

<value><struct><member><name>identity</name><value>5162809890</val

<member><name>password</name><value>foobar</value></membe</struct></value><value><struct><member><name>identity</name><value>5162809895</val<member><name>password</name><value>barfoo</value>

</struct></value></data></array>

</value></member></struct></param></params></methodCall>

CPE

WAN

OSS, Fault Management, etc.

Northbound API

CPE

CPECPE

CPE CPE CPE

Southbound API

WEB GUI DB DB DB

XML-RPC (or SOAP) (GET/SET/EVENTS)

time for telcos to go beyond pots! - uc...

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