WebRTC allows browsers to communicate directly through peer-to-peer connections without plugins. It uses protocols like SRTP for secure media, ICE for network traversal, and SDP for session description. Signaling can be done through any protocol that supports SDP exchange. WebRTC addresses issues like NAT traversal using STUN, TURN servers, and trickle ICE. RTP bundling allows multiple media streams to be multiplexed over a single port.

1. ß
#sthlmwebrtc
http://www.meetup.com/WebRTC-Stockholm/
wifi: sup46-guest-wifi-by-MTGx Password: unfors33N

2. 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.

3. 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

4. Much more than ”calls”
GAMES
DATING SITES
ART

5. Much more than ”media”
DATACHANNELS
p2p (or
browser2browser)
Anything!

6. Secure peer-to-peer
Using DTLS + SRTP for media
Web Server
“Rendevouz”

7. Much more than browsers
Browser
Thing with
WebRTC stack
Gateway
with WebRTC stack
Something else.
SIP, XMPP, DecNet

8. Much more than browsers
App (no browser)
Thing with
WebRTC stack
Gateway
with WebRTC stack
Something else.
SIP, XMPP, DecNet

9. Much more than browsers
SIP phone SIP phone
Gateway
with WebRTC stack
Something else.
SIP, XMPP, DecNet

10. P2P SCTP over UDP
P2P SRTP
P2P SRTP
Multiple channels
AUDIO
VIDEO
DATA
#moreCrypto

11. 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

12. The WebRTC protocols
Defined by the IETF
No signalling protocol
specified
Use SRTP for media
Use ICE (TURN, STUN) for
network discovery

13. 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.

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

15. P2P call setup
with network issues
Web
Server
Media
SDP SDP
The web application defines the signaling
TURN
Server
TURN
Server

16. 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

17. No signalling?
Means ANY
signaling
XMPP
SIP
?
The users need to
exchange SDPs. That’s
all.

18. Media interoperability
OPUS
Audio
ALAW
Audio
H.264 video
VP8 video

19. NAT
Firewall
Network problems

20. A moving target
IPv4
IPv6
IPv4
IPv6
CGN
IPv4
IPv6
Things change during
the session, new interfaces
show up….

21. Using a turn server
TURN
on Internet IP
TURN
on Internet IP

22. Using a STUN server
NAT box
STUN on
Internet IP
Finding
the outside IP on the
NAT box

23. Using a turn server to
bypass firewalls
TURN
on Internet IP
Tunnel
media over TLS/
443

24. Using a turn server to reach
old IPv4
TURN
on Internet IPv6
and IPv4
IPv6 client allocates
IPv4 turn adddress

25. Using a turn server to safely
leave the LAN
TURN
on Internet IP
in DMZ Auto discovery

26. ICE: Finding best media
path
TURN
on Internet IP
Test all
connections, then
start media
STUN: Outside of
NAT IP
1
2
3

27. 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

28. Summary
• Use any signalling you want, as long as you can
exchange SDP documents
• Many ways to handle NAT/Firewalls

29. 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.

30. SIP & WebRTC
Web
Server
SIP server with
web socket
support
SIP phone
Media
SIP Javascript
client

31. SIP: Using gateway
Web
Server
SIP gateway with
web socket
support
SIP phone
Media
SIP Javascript
client

32. SIP: Using gateways
Web
Server
SIP gateway with
web socket
support
SIP phone
SIP Javascript
client
Needs to be able to transcode
Turn
Server

33. Using Open Source
Apache
Web
Server
Asterisk + Opus
Patch
SIP gateway with
web socket
support
Jitsi
SIP phone
JSSIP Javascript
client
ReTurn
Server

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

35. 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

36. 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

37. RTP bundle
The new media stack. All for one. One for all.

38. A RTP session
Exchange of media
Consisting of streams
Each stream has an SSRC and one or more CSRC
Synchronisation source, Contributing source

39. 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

40. SDP declares a stream with
m=
Exchange of media
AUDIO
VIDEO

41. RTP Mux
AUDIO
RTCP
5792
5793
AUDIO
RTCP
5792
Mixing the media stream and RTCP on one port
RFC 5761

42. 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

43. 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

44. For SRTP + SRTCP mux
• Still two encryption streams, multiplex after
encryption

45. RTP Bundle
AUDIO
VIDEO
5792
5793
AUDIO
RTCP
5792
Bundling multiple media streams on one port
VIDEO
RTCP
5804
RTCP 5805

46. 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

47. 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

48. That’s the intro.
Let’s talk!

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