Transcript Intertex Data AB, Sweden

Video-Over-IP
Driving the Need for Internet+
Prepared for:
INGATE’S SIP TRUNK – UC SEMINARS:
SIP Trunking, Video, Collaboration and More
ITEXPO Conference, Miami, February 2012
By:
Karl Erik Ståhl
President Intertex Data AB
CEO and Chairman Ingate Systems AB
[email protected]
Also see:
http://www.ingate.com/files/An_Internet+_Model_for_Global_Unified_Communication.pdf
Whitepaper (in progress)
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
2012Intertex
IntertexData
DataAB
AB
©©2012
1
More on the Internet+ Model
Thursday 2nd, 1:00 pm : Video-Over-IP,
Driving the Need for an Internet+
Friday 3rd, 9:00 am :
BoF, Room A208
Birds-of-a-Feather , Session
Also see:
http://www.ingate.com/files/An_Internet+_Model_for_Global_Unified_Communication.pdf
Internet+ Whitepaper (in progress)
http://www.ingate.com/files/itexpo_miami_2012/Intertex-Overview_of_an_Internet+_Model.pps
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
http://www.ingate.com/itexpo_miami_2012.php
http://www.ingate.com/files/itexpo_miami_2012/Intertex-BoF_Internet+.pps
2012Intertex
IntertexData
DataAB
AB
©©2012
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Intertex & Ingate
 Same parent company
 Intertex: SMB, SOHO and home SIP Firewalls and E-SBCs
• For service provider 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
© 2012 Intertex Data AB
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Telepresence – Why not for Everyone?
 It has been there for years - high end – high cost,
but you save airline tickets.
 But telepresence has more or less rolled out there own wires
 No global connectivity or at best via a certain conference bridge
© 2012 Intertex Data AB
4
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
© 2012 Intertex Data AB
5
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!
© 2012 Intertex Data AB
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Provide Internet+ so we can get Telephony+
 UC rich communication (not just AM radio quality Voice): Bring the
islands (Enterprise UC LAN, Skype, Google Talk and others) together!
 Deliver to the users: On LANs and with Smart Phones!
 UC should be global, with quality and with phone numbers as well as
SIP-addresses!
© 2012 Intertex Data AB
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Internet has Shown the Success of a Cloud!
We need this for global UC:
SIP Connect 1.1
Internet+
With:
1. All you can do with SIP - UC
2. Full mobility
3. Numbers and SIP addresses
4. Quality up to Telepresence
5. Interoperability – Don’t GW,
unless required
6. Delivery to the users
But got this (SDN/IMS):
(Wires on top of the cloud!? Wasn’t
creating the cloud the success?)
Internet
MPLS
Session Delivery Network (SDN) = POTSoIP
It’s Not Even Good for FAXing
 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 and the i3 Forum.
T.38 works fine in one hop!
 And their billing is by voice minutes – Far away from any UC!
 And where did the reliability, scalability and good performance of IP go?
© 2012 Intertex Data AB
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Internet has Shown the Success of a Cloud!
We need this for global UC:
SIP Connect 1.1
Internet+
With:
1. All you can do with SIP - UC
2. Full mobility
3. Numbers and SIP addresses
4. Quality up to Telepresence
5. Interoperability – Don’t GW,
unless required
6. Delivery to the users
But got this (SDN/IMS):
(Wires on top of the cloud!? Wasn’t
creating the cloud the success?)
Internet
MPLS
Session Delivery Network (SDN) = POTSoIP
Provide Internet+ so we can get Telephony+
 UC rich communication (not just AM radio quality Voice): Bring the
islands (Enterprise UC LAN, Skype, Google Talk and others) together!
 Deliver to the users: On LANs and with Smart Phones!
 UC should be global, with quality and with phone numbers as well as
SIP-addresses!
© 2012 Intertex Data AB
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We Are (sometimes) Doing Better!
SoftSwitch/SBC
Overlay
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
Users and Services can be Everywhere: SIP must connect there!
© 2012 Intertex Data AB
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SIP is Self Routing and E-SBCs Can Do it All
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
© 2012 Intertex Data AB
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So, Don’t Just Feed SIP Into POTSoIP…
ONLY FOR
POTS
SoftSwitch/SBC
Overlay
PSTN
UC Voice Mail
Remote
Users
SIParator®
Follow standards so we
don’t need gateways, here,
there and everywhere!
IP-PBX
We do everything else
successfully,
flat over the
Data & VoIP
LAN
Internet. Please let us
have the same for all realtime communication also.
The Internet+ Thanks!
© 2012 Intertex Data AB
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Time for Something Better: Internet+
Learn from the success of the Internet:
Networks shall Not Be Application Specific!
The Internet+:
A non application-specific transport network: Just like the Internet!
It IS the Internet – just extended:
+ Delivery to the users, on LANs and to smart phones
+ Prioritization for real-time traffic - Just enable diffserv
+ Metering and charging of “beyond Internet usage”
Good for everyone:
also for the Telcos:
The SDN is not needed,
Provide something better,
- IP connects end-to-end
- and users will pay for it..
- SIP is a self-routing
 Bill the better!
© 2012 Intertex Data AB
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Time for Something Better: Internet+
Learn from the success of the Internet:
Networks shall Not
Be Application
Specific!
Enable
the new
services,
interoperability
and the standard we
+:
The Internet
and
want!
A non application-specificneed
transport
network:
Just like the Internet!
It IS the Internet – just extended:
the same
time:
+ Delivery to the users,At
on LANs
and to smart
phones
+ Prioritization for real
time traffic
- Justrevenue
enable diffserv
- New
Telco
+ Metering- and
charging
of “beyond
Internet usage”
Vast
Telco
infrastructure
savings
Good for everyone:
How to also
do?for the Telcos:
The SDN is not needed,
Provide something better,
Easier than believed!
- IP connects end-to-end
- and usrs will pay for it..
- SIP is a self-routing
© 2012 Intertex Data AB
 Bill the better!
16
It Should be of Utmost Interest for the Telcos
Internet and Telephony Economy
Internet
+ Economy
Bandwidth Usage
Data
Low Charged
Internet
Bandwidth
Data
Skype etc.
RTC
RTC
Telephony+
Telephony Income (highly charged)
Quality
Bandwidth
New Income
 For real-time usage, we need an Internet pipe with prioritization enabled not just for telepresence quality, but also for 2G, 3G and 4G mobile real-time
usage with smart phones
 Has to be charged separately. If not, it would be used for everything and we are
back at all usage being at the same quality level.
 And we don’t want our smart phone batteries drained
 And we want to use both phone number and addresses – not the many island.
© 2012 Intertex Data AB
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Internet+ Model
The Internet with Quality Enabled
Global IP Transport Network
SIP Connect 1.1
PKI
DNS
EMS
ENUM
TR069
All SIP Routed Everywhere (Not Gatewayed! Via SIP Proxies – Not B2BUAs)
The TOQrouter – Trust, Openness, Quality – is a routing SIP proxy, a billing meter, and with built in SBC.
Quality and Numbers are Important
 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?
© 2012 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
© 2012 Intertex Data AB
Callee
20
SIP is Self Routing and E-SBCs Can Do it All
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
© 2012 Intertex Data AB
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For the Telephony+ Services
For a Telephony+ service (including POTS):
+ SIP is the standard to use. All SIP transported everywhere!
 The SIP interface must be available everywhere and the network carry anything
possible with SIP, both for users and services.
 The Network shall not interfere – not be application specific – that is between users
and services. SIP proxies are allowed, but
 Gateways and B2BUA are only allowed toward outside elements
+ Usage of E.164 numbers in addition to SIP address
+ Telcos must share numbers in a common data base
+ ENUM convert numbers to SIP addresses (and other addresses Skype)
+ Gateways in and out to the other islands.
+ Trust between participants (like having a telephony
subscription/telephone line/number today)
© 2012 Intertex Data AB
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Internet+ Model
The Internet with Quality Enabled
Global IP Transport Network
SIP Connect 1.1
PKI
DNS
EMS
ENUM
TR069
All SIP Routed Everywhere (Not Gatewayed! Via SIP Proxies – Not B2BUAs)
The TOQrouter – Trust, Openness, Quality – is a routing SIP proxy, a billing meter, and with built in SBC.
For the Telcos To Do
 Enable diffserv on Internet+ Accesses
(Or provide separate high quality pipe on routable IP.)
 Provide ENUM directory (public or private)
E.164 numbers to SIP address resolution
 Peer higher quality pipes with other carriers
just as for Internet
Share ENUM between the Carriers
 Deploy TOQrouters* – similar to clever ESBCs used for SIP Trunking
Manage as already done in volume deployments
Provide Certificates to the TOQrouter for trust
TOQ routers use mutual TLS for all WAN SIP
 Pick up CDRs from the TOQrouter and Bill
* TOQ stands for Trust, Openness, Quality
© 2012 Intertex Data AB
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For mobile and Our SmartPhones
Internet+
 Just replace today’s network firewall
with the TOQrouter* and use the IP
channel for good real-time
communication also
  No more battery draining 
(keep-alive packets not needed)
4G 3G 2G
 Forget about VoLTE in 4G networks.
It is POTSoIP again…
 No more ”mobility plumbing”
needed: SIP reaches everywhere!
* TOQ stands for Trust, Openness, Quality
© 2012 Intertex Data AB
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Most Important: SIP Everywhere – Just like HTTP!
We would not have the Web, if HTTP did not go between the Browser and the Web server.
Today SIP is stopped and limited by Firewall/NATs, SoftSwitches and bad SBCs.
The TOQrouter is a standard compliant SIP proxy (and SBC) that routes all SIP between the
Users and Servers according to RCF 3261. The TOQrouter is not interfering with the usage
of the SIP communication (like today’s plumbing), but can measure the usage for billing.
Proper SIP transport (by routing
SIP proxies) is required:
• For all beyond POTS usage, UC
• For eliminating network
incompatibilities – Interop issues
are then reduced to being only
between clients and services
• For mobility: User and services
can be anywhere!
• For global UC: Clients, PBXs,
Cloud services etc. only have to
use a standard SIP interface.
There is no other way to go!
Really Possible? – Don’t we need this?
Will there then be
another 10 years
before Internet+, or?
All standards and
all elements are
ready to use.
No IMS multimedia call across carrier domains after 5
years of deployment. But POTS on RJ11 delivered…
© 2012 Intertex Data AB
And they can be
introduced
step-by-step!
27
The TOQrouter Can Be The Registrar
YOU Shall Decide Whom To Accept Calls From
Example using the Intertex IX78 as TOQrouter:

The TOQrouter is a good registrar, but the registrar can be located anywhere


Your Buddy list and below allows you whom to communicate with
Those on the “Trusted Network” will be the Telephony subscribers as before
Exists on Proxy level and
individual User level
Most of the Gear is Already in Use
…but not (yet) for Internet+
Internet
IP-TV
VoD
VLANs or ADSL
Virtual Circuits
IMS
VoIP
TR-069
WiFi
The Multimedia LAN
IPPBX
PDA
Telepresence
In the above deployment, the
Intertex IX78 E-SBC is used for SIP
trunking, but is actually capable of
TOQrouter functions.
This major European Telco has a
high quality VoIP network using
white addresses and is routed to the
Internet. An Internet+ model would
here simply mean IP peering their
VoIP IP network to other service
providers’ high quality networks,
supplying an ENUM database and
relying on CDRs delivered to the
management system.
The Intertex IX78 already provides
the clean SIP interface to LAN
endpoints and servers on the LAN,
in parallel with its gateway approach
toward the PBX and the IMS system.
The SIP Standard: Global and More Than Voice!
 Today 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… Telcos don’t
like another
Skype. Offer
better and
bill happily
with
Internet+!
© 2012 Intertex Data AB
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Billing – CDRs for Efficient Processing
Now also with Video Call Metrics and Pipe Used!
CDRs with Call Quality Metrics – View from iEMS (our TR-69 management system)
© 2012 Intertex Data AB
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Can the “Core” Soft Switch/SBC Participate?
 Sure - it can be a SIP Registrar - which could be used instead of the registrar
in the TOQrouter (In an Internet+ model, a SIP server can be everywhere!).
 As a routing element; It must be a compliant SIP Proxy (B2BUAs/Gateways
must not be in the transport part of the network)!
 It must only route PSTN calls into the POTSoIP overlay
 It could do some
individual forwarding
etc. of incoming calls
(instead of the
TOQrouter) –IF
behaving like a SIP
Proxy
 The TOQrouter will be
required anyway
© 2012 Intertex Data AB
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Why are there SDN and IMS?
 IMS world said (but could not deliver):
 “Evolving broadband communication by launching new services cost-efficiently“
 “People want an enriched communication experience, anywhere, anytime, and to any device.
By XXX IMS technology operators are able to cost efficiently deliver that experience and to
generate revenue”
 An SDN, but not today’s transport network, the Internet, may achieve:
 service providers can bill for their services,
 the ability to use a higher quality IP transport network,
 the ability to only allow trusted users - that is, subscribers to a service provider - to
participate in the communication,
 fulfilling lawful intercept requirement and
 fulfilling emergency calling requirements.
 The Internet+ model provides the above better, while maintaining:




reliability (no introduction of massive central elements)
scalability (no introduction of massive central elements)
good performance of a global IP transport network
Interoperability (no multiple conversions, no interference with SIP).
© 2012 Intertex Data AB
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The TOQrouter



A Firewall/NAT (with general Firewall security functions)
An RFC 3261 compliant SIP Proxy also implementing RFC 3263 and RFC 3264
The SIP Proxy performing ALG functions by:
- handling and being aware of its NATed environment (by reading the IP routing table)
- reserving NAT ports and rewriting the SDP accordingly (according to the Midcom RFC 3989)
- setting up the NAT and opening pinholes for the media in the Firewall (according to the Midcom RFC 3989)

The SIP Proxy implementing RCF 3325 (trusted networks):
 - using mutual TLS and certificates towards SIP Proxies on the WAN








Having functions for classifying SIP traffic to assign correct QoS class, based on various conditions
A SIP Registrar for (i) keeping and using registrations from LAN connected devices – a Shadow
registrar - to allow incoming calls. This (shadow) registrar should also be able to handle RFC 6140
Gin registration for a PBX. (ii) Being the main registrar for one or several domains.
A function and setup for SIP Domain forwarding to local SIP Servers, e.g. an IP-PBX on the LAN to
be used by remote users.
A dial plan with ENUM look-up to allow E.164 numbers to be used, as described below
QoS based routing, to select correct IP interface, in case special QoS WAN pipes are provided
The TOQmeter– A meter for billing purposes plus trust for the provider
A management interface and protocol, allowing very high network scalability, with trust and
security to allow CDR delivery over a public network (TR-069, sending CDRs in Informs is
recommended.)
The TOQrouter is also the point where a legal requirement of intercept can be fulfilled. And it can
aid emergency calling since its physical location usually is known. (RFC 6442)
© 2012 Intertex Data AB
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The TOQrouter
Optionally, the TOQrouter may include:

Functions in the SIP proxy for improved compatibility towards SIP devices

Gateway functions in a B2BUA for extended compatibility improvements towards non SIP
incompatible devices (e.g. for connecting a variety of PBXs)

Firewall and NAT functions for data traffic

Analog telephone ports (for connecting POTS ports)

Triple play capability, by handling separate IP interfaces for Internet, VoIP/IMS and IP-TV and
VoD etc.

An access modem/router, e.g. for DSL, Cable, GPON, VLAN Ethernet, T1, MPLS

Multimedia capable PBX functionality using the available SIP Proxy and SIP Registrar

Other useful Business and Residential Gateway functions
Notice that these kind of functions must not be confused with, or interfere with, the basic
TOQrouter functions!
© 2012 Intertex Data AB
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What SIP to Use? – Just SIP!
Internet+ does not interfere – Just transports/routes (as HTTP or SNTP)
It is between the endpoints: Between Users and Servers!
For all endpoints using SIP in the Internet+ model, minimum:







RFC 3261 SIP: Session Initiation Protocol
RFC 3263 SIP: Locating SIP Servers – DNS usage, plus
RFC 3264 An Offer/Answer Model with the Session Description Protocol (SDP)
RFC 4028 Timer
RFC 6442 Geolocation header (for emergency calling)
RFC 3325 For endpoints wanting to set Privacy Policies
G.711 codec for minimum voice interoperability
For endpoints wanting confirmed early media (telephones):

RFC 3262 SIP: Prack/100rel for early media
For endpoints using call transfer and similar:



RFC 3515 Refer
RFC 3891 Replaces
RFC 3892 Referred-by
© 2012 Intertex Data AB
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What SIP to Use? – Just SIP!
(continuation)
Internet+ does not interfere – Just transports/routes (as HTTP or SNTP)
For Presence endpoints:



RFC 3265
RFC 3856
RFC 3863
For IM endpoints:

RFC 3428
For servers supporting endpoints (e.g. an IP-PBX) if they want the option of
authenticating their users:


RFC 3325 Asserted Identity within Trusted Networks
RFC 6140 (Gin Registration) or use fix IP ITSP IP address when using SIP Connect 1.1
Extensions, such as (most of?) the IMS additions, will be transported correctly by the
TOQrouter.
© 2012 Intertex Data AB
37
More on the Internet+
Friday 3rd, 9:00 am :
BoF, Room A208
Birds-of-a-Feather , Session
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
© 2012 Intertex Data AB
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