Transcript Intertex Data AB, Sweden

The VoIP Net: From POTS to Quality
Unified Communications Globally
Prepared for:
Ingate Systems 3 Day Seminar
Unified Communications:
SIP Trunking, Video, Collaboration and More
ITEXPO Conference, Austin, September 2011
By:
Karl Erik Ståhl
President Intertex Data AB
CEO and Chairman Ingate Systems AB
[email protected]
Also see Live Demo Presentation from ITEXPO SIP Trunking Summit Miami, February 2011!
http://www.ingate.com/files/ITEXPO_Miami_2011_Presentations/Intertex%20-%20UC%20Across%20the%20Borders.pps
© 2011 Intertex Data AB
Intertex & Ingate
 Same parent company
 Intertex: SMB, SOHO and home SIP Firewalls and E-SBCs
• For volume deployment
 Ingate: Enterprise and SMB SIP Firewalls and E-SBCs
• SIParators® for enterprises and projects
 Cooperation in management and development
 Co-developed SIP code
 Ingate represents Intertex in the US
© 2011 Intertex Data AB
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From POTS to Global Quality UC
 POTS and mobile have global reach, but Low Fi voice only…
 UC is rich communication, but today only in islands (the Enterprise UC
LAN, Skype, Google Talk and the others)
 UC should be global, with quality and with SIP-addresses as well as
phone numbers!
Will it happen?
How?
When?
© 2011 Intertex Data AB
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Yes, Telcos are Concerned About their Core Business!
 Are Telcos just becoming bandwidth providers?
 IP has just been used to replicate POTS Telephony
 Where is the global Live IP Communication: Multimedia or UC?
 The “Beyond POTS” islands are taking over:
• at the Enterprise UC LAN
• by Skype, Google Talk and the others
Why not better and beyond?
Telcos can bring it together and offer better!
© 2011 Intertex Data AB
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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!
We cannot get
stuck here
and not with eight
dect phones either!
Phones can be
More Than Smart? – What about better than AM-radio…
© 2011 Intertex Data AB
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Shouldn’t We be Far Beyond PSTN and POTS?
 POTS and PSTN have been there for 100 years
RJ11
Black
Phone
3.5 kHz
isn’t HiFi,
but MOS
is 5!
 Now we have a new global network: The IP Networks
 And we have a new standard: SIP
RJ45
IP Phone
Soft Client
WiFi Mobile
Presence
LAN
Intranet
Internet
Messaging
© 2011 Intertex Data AB
Voice
Video
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But IP Used to Replace Bits of the PSTN, keeping POTS 
Gateway
US
Gateway
PBX
IP PBX
VPN
Tunnel
Europe
IP PBX
Are we stuck
with old POTS
telephony
over new wires?
Toll
Bypass
PSTN
Gateway
Gateway
Soft
Switch
Very seldom VoIP connectivity
between the VoIP IP clouds!
Voice over
Broadband
Most broadband VoIP providers still run
calls between each other over the PSTN!
© 2011 Intertex Data AB
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SIP wasn’t Meant for Islands or Voice Only!
[email protected]
To receive SIP calls globally:
- A SIP server (Proxy Registrar)
- SIP server domain published in DNS
Proxy Registrar
for partco.com
RING!
DNS
partco.com
Internet
To initiate SIP calls:
- A proxy capable of routing (=DNS lookup!)
- Add ENUM to use E.164 numbers
Outbound proxy
for smartco.com
CALL
[email protected]
Caller
Proxy
Magic? – It’s just the SIP standard…
Callee
Proxy
The SIP tapeziod
Caller
© 2011 Intertex Data AB
Callee
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The SIP Standard: Global and More Than Voice!
 over the Internet, but then:
 not always sufficient quality
 difficult to bill by usage (Telcos’ core business…)
 and the NAT/Firewall traversal issue must be
resolved
 Telcos have feared another Skype… But Telcos
don’t like
another
Skype. Need
to offer more
to bill…
© 2011 Intertex Data AB
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Instead IP and SIP has been Used to Replace Pieces of the PSTN
Maintaining Old Structures  POTSoIP
 We see a telephony overlay structure of Soft Switches and SBCs (or IMS)
on top of IP that is ill suited for anything beyond old time voice.
 And Carriers Peer their Networks PSTN Style…
It is even destructive for the 160 years old Fax service*
* 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
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Telephony Peering: A Show Stopper for Global UC!
 Carrier Peering points are mostly TDM type  POTS only
 There are VoIP Peering – But still
only for Voice Minutes  POTSoIP
 What is a UC Minute? – A dead end!
A price list not even possible…
 Voice, HD Voice, Codec, Video resolution, Video codec, IM, Presence message?
 Quality level used - From Internet to critical telepresence?
 And for new services? The IMS model will not work!
Even if IMS gets it technically working– Global
connectivity will fail due to lack of multimedia price lists
between SPs.
There is STILL NO IMS peering after all these years
 Simple solution:
(only voice peering used).
The SP must only charge its own customers – A great simplification!
 Any charging between carriers should only be IP based – Not application specific!
© 2011 Intertex Data AB
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But We Can Do It and We Are Close!
Qwest
Deutsche Telecom
Internet
MPLS
TeliaSonera Internet
QoS IP Network
QoS IP Network
AT&T
MPLS
MPLS
ENUM
C
D
R
C
D
R
SIParator
IX78
Let’s see:
- What is missing?
- How can we do it?
© 2011 Intertex Data AB
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Missing 1: The Global Quality IP Network
 We have the Internet…
 But you cannot get priority – Not always good enough, e.g. for Telepresence
Qwest
Deutsche Telecom
Internet
TeliaSonera
Internet
AT&T
 We have the Telco’s better VoIP networks
 But they are VoIP islands or (in best case) connected via SBCs for voice peering only
Soft Switch
Soft Switch
Soft Switch
Soft Switch
 We need a higher quality IP network IP Peered globally, the “IQ-Net”
 Just like for the Internet – Settlement free peering (or simple data volume charges)
 For real time traffic only, and with quality bits (DSCP/TOS) honored
IQ-Net = Quality bits (DSCP/TOS) honored
Routed to the Internet will give interoperability – Same network. Here we simply have
access to higher quality. (TeliaSonera’s VoIP network has used that model since long)
© 2011 Intertex Data AB
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Multiple QoS Separated WAN Pipes are Common (Telia Network)
The Multimedia LAN
Internet
IP-TV
VoD
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
Internet
The Multimedia LAN
IPPBX
     
PDA
Telepresence
But only the Internet is IPpeered globally.  IP-peer
the VoIP networks = IQ-Net!
Missing 2: Billing by Usage
 Usage of the better IQ-Net must be billed separately from the Internet
 If not charged separately, it would be used for everything and we are back at all
usage being at the same quality level.
 Service Providers should only charge their customers
 No UC minute charge between carriers. Cannot be defined. – Great simplification!
 Settlement free IP peering has worked for the Internet. Any exchange between
carriers shall only be based on IP level usage (data transported at a certain quality
level).
 Measure usage at certain quality levels = Fair pricing model
 Usage, for each call, voice, video and data transferred and at the quality level used
can be put in CDRs and then billed as usual to the customer.
 Can be done by clever E-SBCs*
* Some operators already require the E-SBC to measure the quality – MOS value – of each call.
The E-SBC can also classify the traffic and assure the correct quality pipe is used.
Such E-SBCs are Telco deployed, Owned and Managed. TR-069 is a secure, highly scalable
management protocol allowing such usage.
© 2011 Intertex Data AB
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3-Party Video Conference with CDRs including Call Quality Metrics
© 2011 Intertex Data AB
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Billing – A Key Thing
Now also with Video Call Metrics and Pipe Used!
CDRs with Call Quality Metrics – View from iEMS (our TR-69 management system)
© 2011 Intertex Data AB
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Missing 3: Delivery to the LAN Users
HTTP, SMTP, etc.
 NAT/Firewall Traversal needs to be handled!
 Has hampered deployment and usage of SIP for 10 years. (Applies
to all such protocols: The need to connect to users on private LANs
and to have media flows on separate ports is not trivial.)
 Skype’s clever traversal was a main cause of its immediate success
 But workaround methods (STUN, TURN, ICE, Far End NAT
Traversal) have their limits. Reliability, battery draining of mobiles,
no QoS (for Internet, not for IQ-Net)
 Handled simply: VoIP services often terminates in analog voice ports
(logically outside the Firewall)  POTS only
 SIP Trunking of PBX’s already often use E-SBCs for NAT traversal
(and more)
SERVER
NAT/FW
designed for
this, but…
FW
SIP (and H.323…) connects Person-to-Person
PERSON
FW
FW
PERSON
Locate the person - Set up a session - Open real time media streams
© 2011 Intertex Data AB
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A CPE is Most Often Required Anyway
 A SIP Proxy Based E-SBC, also routing media, can do it all
 Such are also used for security, SIP normalization,
QoS, failover and features
 And while routing SIP, it can add the
functions found in Soft Switches
 Centralized SBCs doing the NAT traversal, require an individual pipe
(e.g. MPLS or other “VPN”) to the each LAN (security concern also)
 …often ending up in an expensive Telco deployed CPE anyway
 Handling many users, makes it a critical point of failure – Duplication and failover
capability by an E-SBC is often required
Not Missing: PSTN/POTS connectivity
 We have the SIP Trunks
 SIP Trunking of PBXs is really about connecting to PSTN/POTS over IP
 Leave today’s SIP Trunks – Add the UC communication to the same endpoints.
The clever E-SBC can do it.
 For the future – if the operator’s SIP Trunk routes to the IQ-Net – the SIP Trunk
can carry multimedia.
 We have the hosted VoIP services
 Leave them just for PSTN/POTS usage. Clever E-SBC can add the UC
communication to the same endpoints.
 For the future – if the operator’s Soft Switch routes to the IQ-Net – it can carry
multimedia.
© 2011 Intertex Data AB
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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: for NAT/Firewall traversal, PBX interoperability and
Security - But Video is not very different from Voice at this level…
 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
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A SIP Trunking E-SBC May Look Like a Specific “Gateway”!
SIP Trunking
SIP System Provider
PSTN
SIParator®
IP-PBX
Data & VoIP LAN
© 2011 Intertex Data AB
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…but can be an Enabler for SIP Services and Users Everywhere!
SIP Trunking
SIP System Provider
PSTN
UC Voice Mail
Remote
Users
SIParator®
Ingate/Intertex E-SBCs
enable SIP based Live
UC Across the Borders!
(SIP does not traverse
ordinary NAT/Firewalls.)
IP-PBX
Data & VoIP LAN
© 2011 Intertex Data AB
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…and can Today Realize the Global UC Network!
Qwest
Deutsche Telecom
Internet
MPLS
TeliaSonera Internet
QoS IP Network
QoS IP Network
AT&T
MPLS
MPLS
ENUM
C
D
R
C
D
R
SIParator
IX78
© 2011 Intertex Data AB
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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
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And What Is Required to “UC Communicate” Globally?
Talk: SIP!
It is the standard for global live person-to-person communication.
Connect over the IQ-Net - being billed for that usage.
And, the E-SBC can seamlessly integrate*:
Connect over the Internet - probably just flat rate Internet billing.
Connect to the PSTN over the SIP Trunk, being billed as for
POTS voice (or it may be a service on the IQ-Net)
ATA
We don’t need new standards – They already exist!
• But standard deviating endpoints need to adapt or have gateways.
• And minimum requirements for a certain application may of course be
required, e.g. Codec G711 u-law for voice telephony, T.38 for fax, H.26?...
for video etc.
* See: http://www.ingate.com/files/ITEXPO_Miami_2011_Presentations/Intertex%20-%20UC%20Across%20the%20Borders.pps
© 2011 Intertex Data AB
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Can the “Core” Soft Switch/SBC Participate?
 Sure, but since IP peered quality network will be used for the transport
(instead of voice specific telephony peering), the “Soft Switching” is not really
used. But Soft Switches could route calls just like any other SIP Proxy.
 Their role will be more of a hosted service, as the SIP Registrar, and for
applying policies for incoming calls (as an alternative of having it in a local
PBX or in the E-SBC).
 And the E-SBCs will
still be required for
various reason, e.g.
Security,
Interoperability and
(unless a separate
level 2 pipe (”VPN”) is
provided from a central
SBC to each customer)
also for NAT/Firewall
traversal.
© 2011 Intertex Data AB
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Is it coming? The OVCC Initiative (by Polycom)!
A network just for Video Calling or the start of the common global UC network?
Key points:
• A global quality IP network
• Service Providers only
charge their own customers
• SIP is the standard
• SIP addresses (email-like)
and E.164 numbers
© 2011 Intertex Data AB
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E-SBCs & SIP Capable Firewalls
See us at ITEXPO Room 9C!
Intertex Data AB
Ingate Systems Inc.
www.intertex.se
[email protected]
Rissneleden 45
SE-174 44 Sundbyberg
Sweden
sip:[email protected]
Tel: +46 8 6282828
www.ingate.com
[email protected]
7 Farley Road
Hollis, NH 03049
United States
Ph: +1 (603) 883-6569
Tel sv: +46 8 6007750
© 2011 Intertex Data AB
29
E-SBCs & SIP Capable Firewalls
Some extra slides, with further
details follows.
See us at ITEXPO Room 9C!
Intertex Data AB
Ingate Systems Inc.
www.intertex.se
[email protected]
Rissneleden 45
SE-174 44 Sundbyberg
Sweden
sip:[email protected]
Tel: +46 8 6282828
www.ingate.com
[email protected]
7 Farley Road
Hollis, NH 03049
United States
Ph: +1 (603) 883-6569
Tel sv: +46 8 6007750
© 2011 Intertex Data AB
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Going Beyond POTS Means Leaving the
POTSoIP “Session Delivery Networks”
 View it as an improved Internet (driving real time communication)
 With IP level QoS honored
 Not restricted to just flat billing: Also billing by Quality level and Usage
(Access can be billed separately from the Service )
 With delivery of SIP Communication to the LAN (traversing the NAT/Firewall)
 With SIP addresses = E.164 numbers (via ENUM)
 Where a SIP service can be anywhere. The “PBX” or Soft Switch should be the
SIP registrar.
 Where today’s SIP Trunks will be the gateways into the PSTN
 Clever E-SBCs can do it all






NAT/Firewall traversal – delivers SIP to the LAN
Can be the registrar (eases proper SIP addresses: [email protected])
Can be the (multimedia) PBX
Can lookup ENUM and route
Can measure usage and generate CDRs
Can enable access, based on authenticated E-SBC
© 2011 Intertex Data AB
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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 email) for each phone number!
- A usable one like: sip:[email protected] (not [email protected])
Let us have both: +46 8 123456 = [email protected]
And why not the same email and SIP address by default with the subscription?
© 2011 Intertex Data AB
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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
 Soft Switches/SBCs : Building/continuing on 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 or UC 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
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Our CPEs Can even be the PBX – With full UC!
Use with standard SIP clients/phones/terminals. “Federates” with all, globally!


A service provider can also offer the UC PBX (by enabling that
functionality).
Allows an existing PBX installation to be expanded and
updated with SIP clients/phones/terminals for UC
communication and for remote users.
Registrar
Remote Users
PBX with
non-SIP
phones
Soft Client
© 2011 Intertex Data AB
WiFi Mobile
34
And There is Provisioning…
 jkjjk
© 2011 Intertex Data AB
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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.
© 2011 Intertex Data AB
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