Www.netcentrex.NET Enabling Next Generation Networks and Services SIP 2003 Olivier Hersent –...

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Enabling Next Generation Networks and ServicesEnabling Next Generation Networks and Services

SIP 2003

Olivier Hersent – CEO/CTO

2Document under NDA © 2002 NetCentrex S.A. All rights reserved.

AgendaAgenda

NetCentrex overview

What about SIP today ? Class IV domain advanced topics

Routing

Numbering Plan

Class 5 domain advanced topics Protocol discussion

SIP in NetCentrex products


NetCentrex todayNetCentrex today


Office LocationsOffice Locations

NetCentrex, Inc Boston

NetCentrex, S.AParis, Headquarters

Caen, Core Switching R&DLyon, Service Node R&D

Paris, Contact Center R&D

NetCentrex,SL Madrid

NetCentrex, NV Brussels

German sales office

UK sales office

APAC sales office

NetCentrex, Inc San Jose

Italy sales office (Pending)


Integration partners complement our direct presenceIntegration partners complement our direct presence

Hewlett-Packard / Compaq, WW partner presence

Atos Origin, European partner presence

Satec, Spain, Portugal, North Africa & South

America

Anect, Czech Republic, Slovakia

Comptek, Russia

AMT, Russia

CSSI, France

Dimension Data, worldwide partner presence

Coheris, European partner presence

Cap Gemini, WW partner presence

Sema, European partner presence


Key figuresKey figures

160 employees, about 50% in R&D Over 150 customers

Despite the challenging environment, NetCentrex sales grew over 30% in 2002

NetCentrex has reached profitability in 2002

20002001

2002

Since 2000, NetCentrex is one of the fastest growing and most stable next-gen telecom manufacturer


Key Achievements in

2002

Key Achievements in

2002


Customer deploymentsCustomer deployments

Largest worldwide VoIP VPN deployment (Equant) Largest worldwide full VoIP Residential deployment

(Fastweb, 200K users, 1.5M calls/day) Largest worldwide SIP based hosted contact center

(Deutsche Telekom)

Massive deployments of the CAMEL Media Control Server in mobile networks at various service providers worldwide (through HP / Compaq)


SIP in 2002


SIP in 2002SIP in 2002

2002 is the year where things get back to reality:

Failure of major carriers, precisely those with the most ‘advanced’ SIP plans Demand is not so clear, volumes not there. Trials, trials, trials…

Failure of UMTS Basic UMTS pushed back years, advanced VoIP features hardly on the horizon Refocus of UMTS on traditional protocols

Explosion of the telecom bubble, leaving high skepticism on so-called “new applications” Focus on telephony for all major customers Virtually ALL IP-PBXs use H.323 exclusively XP SIP support did not bring any market explosion for carriers… where is the money

? Major changes of RFC 3261 over RFC 2543

Correction of major protocol design errors (PRACK, UPDATE) SIP is getting very complex, but the spec still has a quality problem SIP is controlled by to few companies. Not a healthy situation. Quality comes with

debates and a well defined standardization process.


New Public Network ArchitectureNew Public Network Architecture

Application Domain(Class V or other)

User Access Domain

VoIP Network EdgeDomain

VoIP Class IVDomain

Equipment located in the customer premises, access authentication equipment,

access management equipment

Residential gateway, Cable modem with VoIP, xDSL modem

with VoIP, IP Phone, Access Gatekeeper, SIP Registrar

Collocated equipment for added value applications : IP Centrex, Call Center, etc, and equipment for application usage reporting

VON softswitch, Class V softswitch, Contact Center

Application Server, announcement server, IVR, etc

Collocated equipment for basic and VoIP VPN call routing within

the VoIP network, and infrastructure management

Class IV softswitch, Directory Gatekeeper, Number portability

softswitch, etc

Interfaces with the PSTN and with other VoIP networks,

interdomain call detail reporting, Security equipment

Edge proxy, protocol converter, SS7 gateway, ISDN gateway,

trunking gateway, etc


The Network Edge

Domain


Network DiagramNetwork Diagram

VoIP PSTN GWs

Infra

NetCentrex CCS

E1s

SIP / H.323Network

Class 4/5

Soft IP-Phone

PSTN

SS7 Call Agents

MGCP/T

SS7

SAUSAU

H.323/SIP Phone

H.323 CPE POTS

Voice streams (RTP)

MGCP/L

Core N

W

MGCP CPE

POTS

MGCP IP Phone

H.323/SIP


Key requirements of the SS7 edge functionKey requirements of the SS7 edge function

Must properly map ALL ISUP call scenarios (details follow)

Must be capable of ISUP encapsulation Must detect and implement out-of-band DTMF Must support a standard media flow reroute command (TCS=0,

Re-Invite) Must support the same FAX transport method as the core

network. Must support the voice coders used in the VoIP network Must support loop-back calls

If these requirements are met, regardless of the VoIP protocol (H.323, SIP), ALL telephony services can be built on the layers above


Support of ALL ISUP call flowsSupport of ALL ISUP call flows

Announcements before connect Typical H.323 issue : no faststart or earlyH245 support Typical SIP issue : no PRACK support (reliable provisional

responses)

Announcements on release (Disconnect with PI /release/Release Complete) SIP and H.323 use only ONE release message

CLIR service H.323 must support octet 3a (v2 with extensions, or v3) Vendor specific in SIP, bug in the RFC spec.

Numbering plans, type of number… Not present in SIP, required by many PBXs


Support of ISUP encapsulationSupport of ISUP encapsulation

H.323 already encapsulates almost all Q.931 elements (CLIP/CLIR, Type of Network, Type of Number, etc.)

SIP lacks almost all SS7 / Q931 parameters

H.323/H.246 = SIP/T = encapsulation of the ISUP message, undecoded, as a “black box”

GTD = Cisco proprietary, transport of ISUP decoded, potentially much more powerful (ISUP becomes visible to VoIP components)

For VoIP termination/origination, only Q.931 transparency is required. Full transparency required for SS7 toll bypass only.


Support of media stream redirectionSupport of media stream redirection

A frequently forgotten requirement… unlike the TDM, VoIP routes media streams end to end.

Lack of this feature is a showstopper for all services

CallAgent IP-PBX

Phone A

Phone B

Media GW

MGCP-T

SIP / H.323


Why out-of-band DTMF is important (1), prepaid exampleWhy out-of-band DTMF is important (1), prepaid example

MCSMCS

User ARTP

SIP Call control & signaling

Collect Destination, user Id and PIN code. Validate through WNP procedure

MCSMCS

User A User BRTP

Make call to other end and use Re-Invite to optimize RTP path

MCSMCS

User A User B

INFO msg ’#’

1 2

3 4

RTP

SIP bye

User presses ‘#’. Call is interrupted. IVR sequence restarts. User B disconnected

MCSMCS

User A User CRTP

User wants to call C. Make call to other end and use Re-Invite to optimize RTP path


Why out-of-band DTMF is important (2)Why out-of-band DTMF is important (2)

5 10 20 50 100 200 300One way delay(ms)

0

10

20

30

40

50

TE

LR

(dB

)

30

1% complain about echo problems

10% complain about echo problems

Talker echo tolerance curves (G.131 figure 1)

Echo Cancellation G711 theoretical limitActual VoIP network echo cancellation

Delay with RTP tromboning

Delay without RTP tromboning

CCplatform

CCplatformThese numbers are for average residential

telephony users, and are probably much stricter for professional users, such as call center agents


Out of band DTMF, the #1 issue in SIPOut of band DTMF, the #1 issue in SIP

All H.323 devices must support H.245 UII In SIP, there is no standard !

Most phones still use RTP encoded DTMF (RFC 2833), as well as many network devices

SONUS uses INFO messages, with a payload similar to UII NUERA uses INFO messages, with a MGCP like payload CISCO uses Subscribe/Notify (draft-mahy-sip-signaled-

digits-00.txt) (IOS 12.2.11(T)), or INFO (Cisco specific) New SIP RFC 3265 proposes yet another method

Out-Of-Band DTMF support in SIP is still mostly proprietary


Conclusion on the network edgeConclusion on the network edge

SIP and H.323 are now exact equivalents for telephony: Nothing you can do in H.323 you cannot do in SIP Nothing you can do in SIP you cannot do in H.323

But SIP is still mostly proprietary for many features (ISUP transparency, out-of-band DTMF) OK for pure toll bypass applications, in a SINGLE vendor SS7 Call

Agent environment Creates complex issues in class 5 applications, or any application

linked to the enterprise. Critical choice of SIP enterprise side devices which must support same extensions as core (or use MGCP instead, with SIP conversion in the core network)

Very limited choice of SIP end user devices (most IP-PBX are still H.323 !)


The Class 4 Domain

H.323 and SIP

similarities


H.323 « Light » Class IV issuesH.323 « Light » Class IV issues

Originatinggateway

Direct ModeGK Terminating

gateway

3rd PtyPSTN network

PSTN CO

ARQ 123456789ACF @TGWSETUP 123456789

SETUP 123456789RELEASE (congestion)

RELEASE (congestion)

The Call is lost !But other PSTN parners may have been able

to complete the call.The network doe not improve call failure rate.

Failure rate50%

PerceivedNetworkFailure rate50%


SIP « Light » Class IV issuesSIP « Light » Class IV issues

Originatinggateway

SIPRedirect Server Terminating

gateway

3rd PtyPSTN network

PSTN CO

INVITE 123456789

503 Service UnavailableACK

SETUP 123456789


INVITE 123456789

302 Moved (@TGW)ACK

Failure rate50%

PerceivedNetworkFailure rate50%

The Call is lost !But other PSTN parners may have been able

to complete the call.The network doe not improve call failure rate.


H.323 « True » Class IVH.323 « True » Class IV

Originatinggateway

TerminatingGateway 1…2

3rd PtyPSTN network

PSTN CO

SETUP 123456789RELEASE (congestion)

Now the Call is properly completed.True class IV resolves network congestion cases, both

in the VoIP network and in the PSTN.This allows to peer with less reliable PSTN partners, but

still offer the best call completion rates

CCS

SETUP 123456789SETUP 123456789


SETUP 123456789CONNECT

CONNECT

Failure rate50%Failure rate

50%

Perceived Network Failure rate25%


SIP « True » Class IVSIP « True » Class IV

Originatinggateway

TerminatingGateway 1…2

3rd PtyPSTN network

PSTN CO

SETUP 123456789


CCS

INVITE 123456789 INVITE 123456789

503 Service Unavailable

INVITE 123456789

200 OK200 OK

ACK

ACK ACK

Failure rate50%

Now the Call is properly completed.True class IV resolves network congestion cases, both

in the VoIP network and in the PSTN.This allows to peer with less reliable PSTN partners, but

still offer the best call completion rates

Perceived Network Failure rate25%

Failure rate50%


Protocol convertersProtocol converters

The NetCentrex CCS PTU (Protocol Translation Unit) transcodes H.323 to SIP, and vice versa.

All telephony services are OK (including early media, media rerouting, etc.)

Requires true out-of-band DTMF on SIP side (OK Sonus/Nuera/Cisco)


The Application

Domain : class 5


Class V in VoIP networksClass V in VoIP networks

Two paradigms :

Distributed smart endpoint approach : The endpoint (H.323 or SIP) supports multi-line features (call

waiting, etc), and the softswitch supports single line features (call redirections, accounting, blocking, etc)

This is the favored market approach today for residential (ATA180, most SIP&H.323 phones, etc), except in PAcketCable® environments

Centralized dumb endpoint approach (stimulus) MGCP/L endpoints, Softswitch does all Mainly useful for full Centrex features


NetCentrex approach to class VNetCentrex approach to class V

Because of the modular design of the softswitch Smart endpoint approach available today using the

Subscriber Policy Engine, compatible with a broad range of CPEs

Dumb endpoint approach can be added for selected customers (more IP Centrex type), using the add-on Subscriber Access Unit.


CCSSubscriber Policy Engine

User Access Domain

SIPSIP

Accounting Master Unit

Class 5 Call Control

NetCentrex Allows Native Support of H.323, SIP, and MGCPNetCentrex Allows Native Support of H.323, SIP, and MGCP

SIPRegistrar

SIPRegistrar

CentrelizedLDAP

Database

H.323H.323

SIPSIP

MGCP/LMGCP/L

H.323SIP

H.323SIP

V

AGKAGK

SAUSAU

H.323H.323

H.323SIP

H.323SIP

AccountingBilling

CoreDomain


Which protocol for the

line control ?


CPE control protocols (1)CPE control protocols (1)

H.323 and SIP follow the ISDN philosophy of a “smart phone”.

This approach works well for residential, where the required feature set is local to one subscriber

This approach becomes very challenging when features involve a group of lines (park, pick-up, transfer, etc…), because it is hard to synchronize the state of multiple smart phones.

The PBX market has massively adopted stateless phones for this purpose

MGCP/L follows the stateless/stimulus philosophy


CPE control protocols (2)CPE control protocols (2)

MGCP/L provides the following advantages over H.323 and SIP for advanced class 5 functions:

Comprehensive support of analogue lines Possibility of implementing services on off-hook,

before the call is connected No impact of deployed devices on new services (all

services are 100% centralized) Business Phone Package allows customization of

feature activation buttons, and screen presentation (Hardcoded by the phone manufacturer in SIP and H.323)


Redial PickUpNewCall More

12h50P 02/12/02 650 778 8125

Your current options

Business Phone packageBusiness Phone package


CPEs : Cost, Availability and Advanced ServicesCPEs : Cost, Availability and Advanced ServicesMGCP CPEs SIP CPEs H.323 CPEs

Availability yes yes Yes

Price per line (requested features : 3 way, transfers, basic screen control, user interaction capabilities)

Very tough competition (driven by packetcable demand)

Emerging competition Active competition

Can control analog phones

Yes WeakNo transfer, no off-hook services

WeakNo transfer, no off-hook services

Standards Stable, many vendors (packet cable, NCS)

Not stable (Centrex requires many proprietary messages)

Stable

Service capabilities

All services (100% controlled by network)

Restricted by phone. No “Off-Hook service” capabilities.

No out of band DTMF!

Restricted by phone. No “Off-Hook service” capabilities.


Conclusion on CPE control protocols Conclusion on CPE control protocols

MGCP is more suitable for Centrex services: Full control of MGCP dumb terminals , screen control Off-hook services Rich services for analog phones (connected to CPE MGCP

Gateways: transfer, ) Easier management Easier introduction of new services

Terminal Competition driven by Cable deployments (MGCP/NCS) SIP is still used towards the network (or H.323)

But SIP / H.323 can be used for residential (if no control on off-hook services required)


What about SIP in

NCX products ?


Support for SIP in NetCentrex productsSupport for SIP in NetCentrex products

CCS softswitch SIP is supported since release 2.5 SIP/H.323 conversion is supported since release 3.0

MCS media control server and VXML browser SIP is supported since release 4 All features, including media anti-tromboning, are supported

SAGA hosted contact center SIP is supported in the current release

MyCall® residential application SIP is supported in the current release

VPN application SIP is supported in the current release

IPCentrex® business telephony application SIP is supported in the current release (trunk side, and soho)


SIP Instant MessagingSIP Instant Messaging

• The SIP “SIMPLE” Draft has been implemented, allowing CCS to be used as a routing node for instant messages.• VPN routing and alias transformations are properly handled for instant messages

• Instant messages are taken into account in the CCS CDRs, and can potentially be charged

• The Maestro IN service interface has been extended to support SIMPLE instant messages.

• Multiple modes are offered :• Fully routed mode for enhanced security and complete charging options• Cut-Through mode for enhanced scalability of accounting is not important

Instant Messaging has been fully tested with Microsoft Messenger®, in audio, video and file transfer modes.


PBX Site 2

CTI

Saga800 Contact Center UNEDIC: French unemployment administration

CTI

Customer Relationship

channels

Region1

Region2

AgentWorkstation

This is a killer app for VoIPand SIP !

Region 1

TelephoneNetwork Saga800

Contact Center

TelephoneNetwork

Architecture• 52 virtual call centers with 700 sites total• 1500 simultaneous agents logged on, 8000 total• 15 million calls/year• geographically distributed front-end and back-end architecture PBX Site 1

CTI

MobileMobile

TelephoneTelephone


NetCentrex Allows Native Support of H.323, SIP, and MGCPNetCentrex Allows Native Support of H.323, SIP, and MGCP

CONCLUSION:

Yes, There are good things in SIP !

… but SIP still has too much ‘bubble’ content (Remember WAP ?). A little less marketing and a little more thinking would help SIP become real.SIP still lacks some essential features and robustness.

Forget H.323, and you will forget the 2M+ H.323 ports sold per year (99% of the market). SIP and H.323 will have an equivalent market share only in 2 years.

NetCentrex does not worship protocols, we simply implement them as appropriate according to customer requirements. SIP is not always the best choice.

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