© 2013 Ingate Systems AB 1

© 2013 Ingate Systems AB

Prepared for: ITEXPO Conference, Las-Vegas, August 2013

By: Steven JohnsonPresident Ingate Systems Inc.steve@ingate.com

Also see: http://www.ingate.com/files/An_Internet+_Model_For_Global_Unified_Communication.pdfInternet+ 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

Enabling the Internet for Real-time Communications

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Finally: High Quality Real-Time Communication For Everyone is Approaching

Global real time communication is still POTS (Plain Old Telephony Service) with not even AM radio quality voice

SIP has been used to replicate POTS (It wasn’t meant to be…)

WebRTC has the potential of telepresence quality: Opus HiFi sound and VP8 / H.264 HD video (It is to begin with over the Internet/OTT!)

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Internet

Internet has Shown the Success of a Cloud!We need this for global UC:

With:1. All you can do with SIP - UC2. Full mobility3. Numbers and SIP addresses4. Quality up to Telepresence5. Interoperability – Don’t build

gateways (unless required)6. Delivery to the users

SIP Connect 1.1

Internet+ Sorry, this is POTSoIP(Wires on top of the cloud!? Wasn’t

creating the cloud the success?)

MPLS

Avoid application specific networks (IMS, VoLTE/RCS + plain VoIP=POTSoIP)

Quality Traffic in addition to Best Effort traffic. It is almost there, ready to be used…

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All modern IP networks (including the public Internet) support prioritized traffic (e.g. diffserve or RSVP) – It may just not be enabled or used…

Current Best-Effort data traffic fills our pipes (at the most narrow point, every time we click a webpage or send an email). TCP regulates, IP-packets are lost in the bandwidth sharing mechanism – Even UDP packets carrying real-time data are dropped – IF NOT PRIORITIZED.

Higher bandwidth is not sufficient for beyond telephone-voice applications! (It just reduces the time the pipe is filled – It will still be filled.) Level 4 QoS: use UDP, not TCP is necessary but not sufficient for high quality

Prioritized packets are not lost – Having 30% or even up to 100% of the bandwidth carrying prioritized traffic is OK if just prioritized. (Best-effort traffic just has to wait – retransmitted etc.)

Quality Traffic Over Data Crowded Internet/OTT?

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Less than 200 ms roundtrip delay is fine, beyond 500 ms is really disturbing

Delay is “as good as it can be” with today’s high bandwidth networks. Good routers or SBCs do not buffer large amounts to cause significant delays.

Delays are caused by (double for roundtrip): Speed of light in the fibre (100 ms to the other side of the Earth)

• Watch out for far away turn servers

Jitter buffers at the receiver add 50 - … ms• Don’t convert/transcode every such point may adds a jitter buffer

Narrow access pipes add large jitter. With 100 kbps access, a media packet may have to wait 120 ms for an Ethernet packet in progress (that is why we use 57 bytes ATM over ADSL lines, so prioritized media can come in between).

The packetized media in itself 20 - … ms (We can forget about delays in good routers and SBCs)

How we prioritize is of less importance (diffserve Expedited Forwarding (EF) Assured Forwarding (AF) Class Selector, RSVP etc,). It is prioritization at all versus just Best-Effort data.

And the Delay (Over Any Network)?

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The Internet+ Model

The Internet with Quality Enabled

Our Global IP Transport Network

WebRTC is end-to-end. ICE/STUN/TURN is used through NAT/firewalls

SIP can be routed everywhere. (Not Gatewayed! Via SIP proxies – Not B2BUAs)

The access device including a clever E-SBC can do both.

SIP Connect 1.1

Internet+

We need a “toll to enter the highway” or everyone will chose priority to surf faster – and we will be back to the same priority.

And real-time traffic is more valuable.

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It is Not Far Away: Carriers Have For Long Provided Quality Traffic Over the Broadband

TR-069TR-069InternetInternet

IP-TVVoD

IP-TVVoD

IMSVoIP

IMSVoIP

VLANs or ADSL Virtual Circuits

The Multimedia LAN

WiFi

Telepresence

IP- PBX

But we need the RTC on the LAN

– Not on an RJ11 = POTS

And don’t send our RTC into the POTSoIP structure! – That is a PSTN-gateway. SIP-devices can route to the other endpoint.

RJ11

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A Healthy Win-Win Economy for Users and Carriers

Telephony Income (highly charged)

Low Charged Internet Bandwidth

Data

Limited Quality RTC

SIP, WebRTC = Telephony+

Skype etc.

Bandwidth Usage

Data

RTC

Quality Bandwidth New Income

Now I I I I I I I

E-SBCs with SIP proxies and TURN servers at the carrier demarcation point allow the already available bandwidth to be used for high quality real-time traffic delivery in addition to the best-effort data delivery.

The future loss of income from specific telephone networks , may be replaced by prioritized OTT and Internet traffic, charged separately from less valuable data traffic. The Internet+ model applies to fixed, Wi-Fi and mobile broadband delivery for both SIP and WebRTC traffic.

Decreasing Telephony Income Being Replaced by Real-Time Traffic over Data Crowed OTT and Internet Best Effort Traffic is a Lose-Lose Situation for Both Carriers and Users.

Delivering Prioritized, Separately Charged High Quality Multimedia Traffic Over Existing OTT and Internet Bandwidth, is a Win-Win Solution for Both Carriers and Users

© 2013 Ingate Systems AB 9

See us at the Ingate SIP Trunk-UC Seminars

Ingate Systems Inc.www.ingate.cominfo@ingate.com7 Farley Road Hollis, NH 03049United Statessip:steve@ingate.comTel: +1 (603) 883-6569

Ingate Systems ABwww.ingate.cominfo@ingate.comRissneleden 45 SE-174 44 SundbybergSwedensip:info@ingate.comTel: +46 8 6007750

ITEXPO South Pacific Ballroom J