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Introduction to
WebRTC
Browser to Browser communication
V 2.1 oej@edvina.net 2017-02-15

twitter @oej
Copyright Edvina AB, Sollentuna, Sweden 2014-2017.All rights reserved.
The idea
Browers can play

audio and video. Why not let them use
the webcam and the
microphone?
If so, users can talk to
each other using the
browser.
Using NO PLUGINS
Much more than ”calls”
GAMES
DATING SITES
ART
Much more than ”media”
DATACHANNELS
p2p (or
browser2browser)
Anything!
Secure peer-to-peer
Using DTLS + SRTP for media
Web Server

“Rendevouz”
Much more than browsers
Browser
Thing with

WebRTC stack
Gateway

with WebRTC stack
Something else.

SIP, XMPP, DecNet
Much more than browsers
App (no browser)
Thing with

WebRTC stack
Gateway

with WebRTC stack
Something else.

SIP, XMPP, DecNet
Much more than browsers
SIP phone SIP phone
Gateway

with WebRTC stack
Something else.

SIP, XMPP, DecNet
P2P SCTP over UDP
P2P SRTP
P2P SRTP
Multiple channels
AUDIO
VIDEO
DATA
#moreCrypto
WebRTC API
JavaScript API defined by
the W3C
User needs to allow usage
of camera and
microphone.
Defines the session in a
SDP document
The applications needs to
provide the SDP exchange
between peers
The WebRTC protocols
Defined by the IETF
No signalling protocol
specified
Use SRTP for media
Use ICE (TURN, STUN) for
network discovery
WebRTC media
Can be bundled and
mixed to one single
RTP port
Always use secure
RTP with DTLS key
exchange
Use ICE to find the
best media path
Use TURN to traverse
NAT and firewalls.
P2P call setup
Web

Server
Media
SDP SDP
The web application defines the signaling
P2P call setup

with network issues
Web

Server
Media
SDP SDP
The web application defines the signaling
TURN

Server
TURN

Server
P2P call setup

with Changing networks
Wifi
Media
If a device moves from one network

to another during a session,
a turn server can handle session

handover to the new network.
TURN

Server
TURN

Server
4G
No signalling?
Means ANY

signaling
XMPP
SIP
?
The users need to
exchange SDPs. That’s
all.
Media interoperability
OPUS
Audio
ALAW
Audio
H.264 video
VP8 video
NAT
Firewall
Network problems
A moving target
IPv4
IPv6
IPv4
IPv6
CGN
IPv4
IPv6
Things change during
the session, new interfaces
show up….
Using a turn server
TURN

on Internet IP
TURN

on Internet IP
Using a STUN server
NAT box
STUN on
Internet IP
Finding
the outside IP on the
NAT box
Using a turn server to
bypass firewalls
TURN

on Internet IP
Tunnel
media over TLS/
443
Using a turn server to reach
old IPv4
TURN

on Internet IPv6
and IPv4
IPv6 client allocates
IPv4 turn adddress
Using a turn server to safely
leave the LAN
TURN

on Internet IP
in DMZ Auto discovery
ICE: Finding best media
path
TURN

on Internet IP
Test all
connections, then
start media
STUN: Outside of
NAT IP
1
2
3
Trickle ice: Start media early,
find best later
TURN

on Internet IP
STUN: Outside of
NAT IP
1
2
3
Start media
where a connection
is found
Then spend
time to find BEST
connection
Summary
• Use any signalling you want, as long as you can
exchange SDP documents
• Many ways to handle NAT/Firewalls
Compatibility with SIP
WebRTC use a lot of updates to
SDP and RTP.
Most old SIP clients and servers
does not directly support it.
A gateway will unfortunately 

be needed.
SIP & WebRTC
Web

Server
SIP server with
web socket
support
SIP phone
Media
SIP Javascript

client
SIP: Using gateway
Web

Server
SIP gateway with
web socket
support
SIP phone
Media
SIP Javascript

client
SIP: Using gateways
Web

Server
SIP gateway with
web socket
support
SIP phone
SIP Javascript

client
Needs to be able to transcode
Turn

Server
Using Open Source
Apache

Web

Server
Asterisk + Opus
Patch

SIP gateway with
web socket
support
Jitsi

SIP phone
JSSIP Javascript

client
ReTurn

Server
The moving target problem
• Early implementations work to follow standard
• A push towards #MoreCrypto and better crypto
• Browsers push the gateways
• Continuous implementation issues
The moving target problem

- Protocols
• Early implementations work to follow standard
• A push towards #MoreCrypto and better crypto
• Browsers push the gateways
• Continuous implementation issues
The moving target problem

- API’s
• Early implementations work to follow standard
• Implementing private name space to indicate
early proprietary implementation
• Applications mix name spaces, have
interoperability issues
• Continuous implementation issues
RTP bundle
The new media stack. All for one. One for all.
A RTP session
Exchange of media
Consisting of streams
Each stream has an SSRC and one or more CSRC
Synchronisation source, Contributing source
A RTP stream
Exchange of media
Consisting of streams
Each stream has an SSRC and one or more CSRC
Synchronisation source, Contributing source (Random numbers)
RTPRTCP
RTCP gives a SSRC a CNAME as an identifier
SDP declares a stream with
m=
Exchange of media
AUDIO
VIDEO
RTP Mux
AUDIO
RTCP
5792
5793
AUDIO
RTCP
5792
Mixing the media stream and RTCP on one port
RFC 5761
Issues with RTP+RTCP mix
• RTP payload types 64-65, 72-80 conflicts with RTCP
packet types
• RTP payload types 64-95 should not be used,
dynamic payload types should be in the range 96-127
when possible
• RTP Payload types below 64 can be used, if not
conflicting with statically assigned numbers
• RTCP packet types 1-191 and 224-254 should not be
used
Example SDP
v=0
o=csp 1153134164 1153134164 IN IP6 2001:DB8::211:24ff:fea3:7a2e
s=-
c=IN IP6 2001:DB8::211:24ff:fea3:7a2e
t=1153134164 1153137764
m=audio 49170 RTP/AVP 97
a=rtpmap:97 iLBC/8000
a=rtcp-mux
This device is capable of receiving
RTCP-mux
For SRTP + SRTCP mux
• Still two encryption streams, multiplex after
encryption
RTP Bundle
AUDIO
VIDEO
5792
5793
AUDIO
RTCP
5792
Bundling multiple media streams on one port
VIDEO
RTCP
5804
RTCP 5805
Bundle
• Use sdp and RTCP to handle bundling
• Define a SDP grouping scheme called BUNDLE
• Define an RTP header extension
• One SDP can contain multiple SDP bundles
• Multiple media flows in one RTP session
Bundle offer/answer
• Offer contains bundle
• Answer may not contain BUNDLE and then
streams are separated
• If answer includes BUNDLE streams are
bundled
• Some streams may not work without bundle
That’s the intro.
Let’s talk!
Edvina Training classes
CUSTOM

INHOUSE

TRAININGS
• Update for existing
Kamailio, OpenSER
and SER users
• Introduces Kamailio
version 4
• New SIP standards
• SIP, Asterisk, Kamailio
and much more
• Done at your site,
customized for your
project
• Cost effective when
more than four
students
Find more details at http://edvina.net

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