Page 3 - Skype Buzz Spring 2016
P. 3
2nd Quarter 2016
SkypeBuzz by Telecom Reseller 3
SHERWIN SIM, CONTINUED FROM PAGE 1 BARRY SPIELMAN, CONTINUED FROM PAGE 1
made the decision to build a business around
WEBRTC IN THE FUTURE
invested in video quality and media processing legacy equipment, to the current functionality
making WebRTC as easy to use as possible. As Vidyo and Google continue to work together, improvements over the last year or two, WebRTC gap between the on-premises and cloud-based
With WebRTC’s evolution over the past couple having an “encode once but send multiple leverages the best open source codecs for video o erings and even objective availability and
of years, and in part because of what we’ve done resolutions” type codec will help developers and audio, and there are mechanisms available
regulatory requirements which may mean that
with Skylink, our platform as a service (PaaS), adjust codec resolution dynamically. ere are to help manage video and audio quality in Skype for Business Online may not be available
and to another extent, our WebRTC Plug-in
many advantages to supporting an SVC codec as WebRTC-powered applications. ere will be at various locations around the world for the
for Internet Explorer and Safari, there is a lot of adjustments will not require a renegotiation or re- more improvements coming soon, and we’re foreseeable future.
interest and more questions that we receive on encode or codecs. e browser’s current WebRTC excited about the future.
Planning the Migration to the Cloud
a regular basis about how Skype and WebRTC implementations do not have a SVC compatible As technology advancements make the leap As such, in almost any scenario in the
compare in other ways.
codec built into it. When this happens, I’m excited into real world implementations, we will see coming years, most enterprises will likely be
to see codec support for SVC regardless if it is video experiences that adapt very well to the implementing a migration strategy from a legacy
WEBRTC, SKYPE AND VIDEO VP9, H.264 or HVEC (H.265).
reality of best-e ort networks. WebRTC will TDM or IP-based PBX system to Microso UC,
QUALITY
ultimately provide the best experience for given as for all but the smallest of organizations, a
e other day a customer asked me about how SO WHAT DOES THIS MEAN?
network conditions, by leveraging the best video full switchover is simply not practical. A smart
WebRTC and Skype compare in terms of video Currently, WebRTC-enabled applications such and audio codecs as part of the standard.
approach would be to gradually migrate workers
quality, so I thought I’d take a few minutes and as Facebook Messenger and Google Hangouts, We’ll discuss other aspects of how WebRTC who can bene t from the cloud today while
write a short post about that.
or any application powered by our own Skylink compares to existing applications and keeping other workers who need the full feature
Having been in the real-time communication Platform as a Service do provide a good technologies like Skype, going forward, but I set or have other reasons requiring on-premises
industry for over a decade, I’ve been a Skype user communication experience. ere are other hope this is helpful to people who are wondering PSTN connectivity as described above, on the
since the beginning. We have to begrudgingly areas where we think WebRTC has massive how the two di er in terms of managing video
admit that, lately, Skype has been doing a good advantages over Skype. And while Skype has
quality. n
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job at providing audio and video services over
“best e ort” networks, aka e Internet.
Skype has put in an enormous amount of
engineering in their communication protocols,
their audio video encoders/decoders, and their
ability to manage a quality experience given the
available network conditions a consumer may
have access to. Just looking at video quality for
example, when a wideband connection starts to
be constricted into a narrow-band connection,
Skype excels at being able to manage that
experience by adjusting the bitrate, frame rate,
and video resolution.
ese parameters are the key knobs to adjust
when managing quality. Allowing users to
maintain continuity in their conversations with
audio/video adjusting appropriately is important
to making video pervasive and accessible.
e end result? ere is a reason why so many
people use Skype as a fallback when other
communications services and applications just
don’t work. Still, it’s a standalone application and
even with Microso ’s e orts to get developers
to “transform their solutions with integrated
communications and the power of Skype”,
they’re a long way from making it as easy and
powerful as WebRTC can to enable contextual
communications.
A comparison between WebRTC and Skype
media quality has to take into account the
speci c use case. For the sake of simplicity. let’s
focus on delivering media between just two peers
or “callers”. e ability to manage bandwidth for
multi-party calls with more than just a couple or
a few peers is more complicated (and more fun).
We’ll save that for other discussions.
WEBRTC AND ITS CURRENT
IMPLEMENTATION
e major web browsers out there each provides
some basic capability related to bandwidth
management. If you’re building applications
with real-time communication features powered
by WebRTC, you have some ability to set the
resolution and bandwidth constraints for video
and audio interactions, to manage the experience
for your users. e core WebRTC engine does
have built in mechanisms to provide feedback
between media streams (RTCP-FB) and it will
attempt to correct and adjust bit-rates in response
to bandwidth quality. And some browsers have
also allowed the ability to reo er SDPs with
adjustments to the bandwidth attribute (although
you may have to modify them yourself).
WEBRTC 1.0 STANDARDS CHANGES
e W3C have certainly heard that adjusting the
target quality/bandwidth is something developers
do want to do while a stream is active. e ability
to massage RTP sending parameters, speci cally
degradation preferences, priority, framerate, bitrate
and resolution scale have been plumbed in.
Not only will this allow WebRTC developers
to be able to change some parameters, but it will
also future proof us for codec advancements like
scalable video coding (SVC).