Implementation Lessons using WebRTC in Asterisk

Implementation Lessons using
WebRTC in Asterisk
Astricon, October 2013
Moisés Silva <moy@sangoma.com>
Manager, Software Engineering

Agenda

Sangoma Technologies - © 2013

• WebRTC Intro
• WebRTC Asterisk Architecture

• Install & Config
• Troubleshooting

2

WebRTC Intro


• It is not a phone in the browser!

3

WebRTC Intro


• It is a full RTC engine in the browser!

4

WebRTC Intro


• Yes, it can be used for a phone in the browser 

5

WebRTC Intro


• Full media engine API in the web browser
• No “call” or “session” signaling defined

• Generic data interchange between browsers,
peer to peer
• State of the art NAT traversal techniques

6

WebRTC Intro


• WebRTC comes with multiple APIs, ie:
• Peer-to-Peer Connections (RTCPeerConnection)
• Peer-to-Peer Data API (RTCDataChannel)
• Statistics (RTCStats)

• Media Stream (getUserMedia)

7

WebRTC Intro


• WebRTC uses established protocols:
• SRTP/SRTCP for media exchange (secure RTP)
• SDP (its use is controversial and currently challenged)
• ICE, STUN, TURN for NAT Traversal
• DTLS for key exchange
• G.711, Opus, VP8/H.264 etc; for voice and video
8

WebRTC Intro


• What signaling to use is up to you:
• SIP
• XMPP/Jingle
• RESTful API (json)

• OpenPeer ….

9

WebRTC Intro


• Applications
•
•
•
•

A phone, video calls, conferencing etc!
Video games
P2P Video Streaming (Chromecast)
Motion-detecting Baby Monitor
(https://github.com/webrtcHacks/webrtc_baby_mon
itor)

10


WebRTC Intro
• WebRTC Web Triangle
Web

Signaling

Signaling

Encrypted Media

Alice’s Browser

Bob’s Browser

11


WebRTC in Asterisk

SIP over WS

SIP over WS
Encrypted Media

Encrypted Media

Alice’s Browser

Bob’s Browser

12

WebRTC in Asterisk


• WebRTC Gateway
SIP/RTP, Jingle, FXO/FXS,
PRI, SS7 etc …

SIP over WS
Encrypted Media

Alice’s Browser

13


WebRTC in Asterisk

Javascript SIP

res_http_websocket

chan_sip

WebRTC

res_rtp_asterisk

res_srtp

14


WebRTC in Asterisk

sipml5
Asterisk 11
Chrome 30

15

Installing WebRTC Support


• Make sure you have:
• libuuid-devel (required by res_rtp_asterisk)
• OpenSSL w/ DTLS support (1.0.1e has
SSL_CTX_set_tlsext_use_srtp)
• libsrtp-devel

16

Installing WebRTC Support


• Easy usual steps …
• ./configure
• make menuselect:
• res_http_websocket
• res_rtp_asterisk

• make install

17

Configuring WebRTC Support


• Enable the websockets server (http.conf)
• enabled=yes
• bindaddr=0.0.0.0
• bindport=8088

18



• Good idea to use secure websockets (http.conf)
• tlsenable=yes
• tlsbindaddr=0.0.0.0:8089
• tlscertfile=localhost.crt
• tlsprivatekey=localhost.key

19



• But … Asterisk currently seems to have issues
with secure WebSockets, patches available to fix
them
• https://issues.asterisk.org/jira/browse/ASTERISK21930
• http://svnview.digium.com/svn/asterisk/team/moy/
webrtc-11/

20



• Verify the HTTP server status

21



• Test websockets connectivity
• npm install –g ws
• wscat –s echo –c ws://<host>:<port>/ws
wscat –s echo –c wss://<host>:<port>/ws

22



• Test websockets connectivity

23



24



• Enable SIP over websockets (sip.conf)
• transport=udp,ws,wss
• Make sure you use the /ws URL when connecting
from JavaScript

25



• Testing using sipml5.org/call.htm

26



27

Troubleshooting


• Troubleshooting Toolkit
• javascript console
• chrome://webrtc-internals
• Node ws (test websockets)

• Wireshark!

28

Troubleshooting


• The javascript console is your friend

29

Troubleshooting


• Checking out chrome://webrtc-internals

30

Troubleshooting


• Checking out chrome://webrtc-internals

31

Troubleshooting


• Note that Wireshark VoIP calls menu won’t work
for calls over websockets
• You can however use “Follow TCP stream” and
see the entire SIP signaling flow
• RTP decoding will not work either (rtcp-mux)

32

Troubleshooting


• TLS decryption can be achieved by installing the
private key on Wireshark
• Preferences -> Protocols -> SSL -> RSA Key List

33

Troubleshooting


• Wireshark decrypted secure WebSocket payload

34

Future


• Trickle Ice
• Use of other codecs (ie Opus, iSAC)

• Video (VP8)
• Use libwebsockets in res_http_websocket?

35

Conclusion


• Asterisk + WebRTC gateway is easy to setup!
• Know your debugging tools

• Understand the protocols involved
• Have fun and hack away!

36


Contact Us
• Sangoma Technologies
100 Renfrew Drive, Suite 100
Markham, Ontario L3R 9R6
Canada

• Website
http://www.sangoma.com/

• Telephone
+1 905 474 1990 x2 (for Sales)

• Email
sales@sangoma.com

38

Implementation Lessons using WebRTC in Asterisk

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