2 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!)
3 Internet Internet has Shown the Success of a Cloud! We need this for global UC: 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 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…
4 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 ). 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?
5 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)?
6 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.
7 It is Not Far Away: Carriers Have For Long Provided Quality Traffic Over the Broadband TR-069 Internet IP-TV VoD IP-TV VoD IMS VoIP IMS VoIP VLANs or ADSL Virtual Circuits The Multimedia LAN WiFiWiFi 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
8 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 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