The Future of SIP in WebRTC
- 1. The Future of SIP in WebRTC
Real Time Communica.ons on the Web
BDNOG5
SHAILA SHARMIN
LINK3 TECHNOLOGIES LTD
DHAKA, 11 APRIL 2016
- 2. WebRTC : Web Real Time CommunicaLons
What WebRTC is :
Browsers with interac.ve voice &
video communica.ons And data
sharing capabili.es
No download or plug-in
• Easy to write audio/video service
• Communica.ons become a
feature, not the applica.on
• Purpose built for the World Wide
Web
• Collabora.ve W3C and IETF
standardiza.on
• Peer 2 Peer
- 5. SIP and WebRTC are different in their
funcLons?
SIP stands for Session Ini.a.on Protocol,
it is a text-based protocol used in
Internet telephony (VoIP) for signaling
and controlling mul.media sessions.
It’s like the square and rectangle concept; all squares
are rectangles, but not all rectangles are squares. SIP
can exist without WebRTC, but WebRTC needs a
signaling protocol to fully operate.
The WebRTC vs. SIP baYle is actually a set
of two different baYles going on at once:
1. SIP vs. Signaling Protocol X
2. WebRTC vs. VoIP (Browser vs. PSTN)
SIP does signaling. And also defines
how media gets handled. WebRTC
does media. But WebRTC doesn’t
define how signaling is handled. Nor
does it care.
- 6. Voice Over Internet Protocol
§ “VoIP” a Broad term
§ Grown to encompass
mul.media, not just voice
§ Diverse protocols
§ some well defined
standards, some de-facto,
some proprietary
§ Used in a variety of
networks
§ IPv4, IPv6, Public Internet,
Private LANs, etc.
UC
Business
VoIP
Fixed Line
- 7. Interworking with tradiLonal VoIP
IPv4 / IPv6
Network
Media
Transport
Media
Descrip.on
Signaling
Transport
Signaling
Protocol
DTLS-SRTP
STUN
ICE TURN
RTP-Mux RTP
BUNDLE
Data
Channels
MSID
UDP SDES-SRTP
MSRP
Unique Transport per Stream
Tradi.onal
SDP
TCP TLS UDP
WebSockets
HTTP
Undefined
SIP XMPP
H.323
Codecs
Opus
VP8
H.264
AMR-WB
VoIP and WebRTC Similari.es
• Transmission of communica.on
data between Users in real-.me
• Use RTP, SDP O/A
• G.711
• Run over IP networks
VoIP and WebRTC differences
VoIP uses a mul.tude of
variants such as VoIP over DSL/
cable modem, VoWiFi/3G,
VoLTE, and Rich Communica.on
Suite (RCS), while WebRTC is
focused on browser-based
communica.ons.
- 8. Let's Talk Signaling- WebRTC does not define a signaling protocol
But signaling is required for call setup,
WebRTC solu.ons must include a signaling
server. WebRTC itself doesn't care how that
server implements signaling, but it must exist
somewhere in the network -- which brings us
to SIP.
SIP defines signaling. Session Descrip.on
Protocol (SDP), defines media. SIP and SDP
work together to create, manage, and tear
down media sessions of any type.
Signaling was lek out of WebRTC for two good
reasons:
1. Different applica.ons may require/prefer
different protocols. The WebRTC working
group did not want to lock it down to
something that may turn out to be inadequate
for all its uses.
2. WebRTC runs in a Web browser, and
support for signaling would require that Web
pages be stateful. This becomes problema.c if
signaling is lost each .me a page reloads.
- 9. Three aspects of WebRTC that marginalizing
the importance of SIP!
1. WebRTC is all about “dumbing down” communicaKons – making it accessible to a lot more
developers than just us VoIP engineers.
2. WebRTC is about embedding communicaKons – changing it from a service into a feature of
another service.
3. WebRTC is about killing federaKons – WebRTC is en.cing a silo approach to services. You
need comms? Just plug WebRTC in and you’re done. No need to think about interworking
with others, connec.ng or federa.ng with more networks .