Transcript IPCentrex Equant Presentation

Enabling Next Generation Networks and Services
SIP 2003
Olivier Hersent – CEO/CTO
www.netcentrex.NET
Agenda
 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
Document under NDA © 2002 NetCentrex S.A.
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2
NetCentrex today
Document under NDA © 2002 NetCentrex S.A.
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3
Office Locations
NetCentrex, Inc San Jose
NetCentrex, Inc Boston
UK sales office
German sales office
NetCentrex, NV Brussels
NetCentrex, S.A
Paris, Headquarters
Caen, Core Switching R&D
Lyon, Service Node R&D
Paris, Contact Center R&D
APAC sales office
Italy sales office (Pending)
NetCentrex,SL Madrid
Document under NDA © 2002 NetCentrex S.A.
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4
Integration partners complement our direct presence
Hewlett-Packard / Compaq, WW
partner presence
Cap Gemini, WW partner
presence
Dimension Data, worldwide
partner presence
Atos Origin, European
partner presence
Sema, European partner
presence
Coheris, European partner
presence
Comptek, Russia
AMT, Russia
Anect, Czech Republic,
Slovakia
CSSI, France
Satec, Spain, Portugal,
North Africa & South
America
Document under NDA © 2002 NetCentrex S.A.
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5
Key figures
 160 employees, about 50% in R&D
 Over 150 customers
2002
2001
2000
 Despite the challenging environment, NetCentrex
sales grew over 30% in 2002
 NetCentrex has reached profitability in 2002
Since 2000, NetCentrex is one of the fastest growing
and most stable next-gen telecom manufacturer
Document under NDA © 2002 NetCentrex S.A.
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6
Key Achievements in
2002
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7
Customer 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)
Document under NDA © 2002 NetCentrex S.A.
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8
SIP in 2002
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9
SIP 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.
Document under NDA © 2002 NetCentrex S.A.
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10
New Public Network Architecture
Equipment located in the
customer premises, access
authentication equipment,
access management equipment
User Access Domain
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
Application Domain
(Class V or other)
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
VoIP Class IV
Domain
Class IV softswitch, Directory
Gatekeeper, Number portability
softswitch, etc
Interfaces with the PSTN and
with other VoIP networks,
interdomain call detail reporting,
Security equipment
VoIP Network Edge
Domain
Edge proxy, protocol converter,
SS7 gateway, ISDN gateway,
trunking gateway, etc
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11
The Network Edge
Domain
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12
Network Diagram
Class 4/5
NetCentrex CCS
Soft
IP-Phone
SS7
Call Agents
H.323/SIP
Infra
SS7
SAU
MGCP/T
SIP / H.323
Network
POTS
MGCP CPE
PSTN
Voice streams (RTP)
E1s
VoIP PSTN
GWs
MGCP IP Phone
H.323 CPE
POTS
H.323/SIP Phone
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13
Key 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
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14
Support 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
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15
Support 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.
Document under NDA © 2002 NetCentrex S.A.
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16
Support 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
Call
Agent
SIP / H.323
IP-PBX
Phone A
Media GW
Phone B
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17
Why out-of-band DTMF is important (1), prepaid example
2
1
MCS
MCS
SIP Call control
& signaling
User A
RTP
User A
Collect Destination, user Id
and PIN code. Validate
through WNP procedure
3
User B
RTP
Make call to other end and
use Re-Invite to optimize
RTP path
4
INFO msg ’#’
MCS
MCS
SIP bye
User A
RTP
User B
User presses ‘#’. Call is
interrupted. IVR sequence
restarts. User B disconnected
User A
User C
RTP
User wants to call C. Make
call to other end and use ReInvite to optimize RTP path
Document under NDA © 2002 NetCentrex S.A.
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18
Why out-of-band DTMF is important (2)
TELR (dB)
1% complain about echo problems
50
Echo Cancellation G711 theoretical limit
Actual VoIP network echo cancellation
10% complain about echo problems
40
Delay with RTP tromboning
30
CC
platform
20
10
05
10
20 30
50
100
200 300One way delay
(ms)
Delay without RTP tromboning
Talker echo tolerance curves (G.131 figure 1)
These numbers are for average residential
telephony users, and are probably much stricter for
professional users, such as call center agents
Document under NDA © 2002 NetCentrex S.A.
CC
platform
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19
Out 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-signaleddigits-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
Document under NDA © 2002 NetCentrex S.A.
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20
Conclusion 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 !)
Document under NDA © 2002 NetCentrex S.A.
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21
The Class 4 Domain
H.323 and SIP
similarities
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22
H.323 « Light » Class IV issues
Originating
gateway
Direct Mode
GK
Terminating
gateway
3rd Pty
PSTN
network
PSTN CO
ARQ 123456789
ACF @TGW
SETUP 123456789
Perceived
Network
Failure rate
50%
RELEASE (congestion)
SETUP 123456789
RELEASE (congestion)
Failure rate
50%
The Call is lost !
But other PSTN parners may have been able
to complete the call.
The network doe not improve call failure rate.
Document under NDA © 2002 NetCentrex S.A.
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23
SIP « Light » Class IV issues
SIP
Redirect Server
Originating
gateway
INVITE 123456789
Terminating
gateway
302 Moved (@TGW)
ACK
INVITE 123456789
Perceived
Network
Failure rate
50%
3rd Pty
PSTN
network
PSTN CO
SETUP 123456789
503 Service Unavailable
RELEASE (congestion)
ACK
Failure rate
50%
The Call is lost !
But other PSTN parners may have been able
to complete the call.
The network doe not improve call failure rate.
Document under NDA © 2002 NetCentrex S.A.
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24
H.323 « True » Class IV
Originating
gateway
CCS
3rd Pty
PSTN
network
PSTN CO
Terminating
Gateway 1…2
SETUP 123456789
SETUP 123456789
Perceived
Network
Failure
rate
25%
SETUP 123456789
RELEASE (congestion)
RELEASE (congestion)
SETUP 123456789
CONNECT
CONNECT
Failure rate
50%
Failure rate
50%
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
Document under NDA © 2002 NetCentrex S.A.
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25
SIP « True » Class IV
Originating
gateway
INVITE 123456789
3rd Pty
PSTN
network
PSTN CO
Terminating
Gateway 1…2
CCS
INVITE 123456789
SETUP 123456789
503 Service Unavailable RELEASE (congestion)
Perceived
Network
Failure
rate
25%
ACK
INVITE 123456789
200 OK
ACK
200 OK
ACK
Failure rate
50%
Failure rate
50%
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
Document under NDA © 2002 NetCentrex S.A.
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26
Protocol 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)
Document under NDA © 2002 NetCentrex S.A.
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27
The Application
Domain : class 5
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Class 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
Document under NDA © 2002 NetCentrex S.A.
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29
NetCentrex 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.
Document under NDA © 2002 NetCentrex S.A.
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30
NetCentrex Allows Native Support of H.323, SIP, and MGCP
Class 5 Call Control
User Access Domain
Accounting
Billing
H.323
AGK
H.323
Accounting Master Unit
V
SIP
SIP
Registrar
SIP
H.323
SIP
Core
Domain
CCS
Subscriber
Policy Engine
MGCP/L
SAU
H.323
SIP
Centrelized
LDAP
Database
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31
Which protocol for the
line control ?
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32
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
Document under NDA © 2002 NetCentrex S.A.
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33
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)
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34
Business Phone package
12h50P 02/12/02
650 778 8125
Your current options
Redial PickUpNewCall More
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35
CPEs : Cost, Availability and Advanced Services
MGCP CPEs
SIP CPEs
H.323 CPEs
Availability
yes
yes
Yes
Price per line
Very tough
competition (driven by
packetcable demand)
Emerging competition
Active competition
Can control
analog phones
Yes
Weak
Weak
No transfer, no offhook services
No transfer, no offhook services
Standards
Stable, many
vendors
Not stable
Stable
(requested features : 3
way, transfers, basic
screen control, user
interaction capabilities)
(packet cable, NCS)
Service
capabilities
All services
(100% controlled by
network)
(Centrex requires
many proprietary
messages)
Restricted by phone.
No “Off-Hook service”
capabilities.
No out of band
DTMF!
Restricted by phone.
No “Off-Hook service”
capabilities.
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36
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 offhook services required)
Document under NDA © 2002 NetCentrex S.A.
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37
What about SIP in
NCX products ?
Document under NDA © 2002 NetCentrex S.A.
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38
Support 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)
Document under NDA © 2002 NetCentrex S.A.
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39
SIP 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.
Document under NDA © 2002 NetCentrex S.A.
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40
Saga800 Contact Center
UNEDIC: French unemployment administration
Customer
Relationship
channels
Telephone
Mobile
Region2
Telephone
Network
Saga800
Contact Center
Telephone
Network
Region1
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
CTI
Region 1
PBX
Site 2
Site 1
CTI
This is a killer app for VoIP
and SIP !
Agent
Workstation
Document under NDA © 2002 NetCentrex S.A.
CTI
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41
NetCentrex 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.
Document under NDA © 2002 NetCentrex S.A.
All rights reserved.
42