WEBRTC : EXPLORATION THROUGH THE QUESTION OF INTEROPERABILITY WITH SIP

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WEBRTC : EXPLORATION THROUGH THE QUESTION

OF INTEROPERABILITY WITH SIP

Soutenance

17/06/2013

Ornella Annicchiarico, Benoit Le Quéau, Mouhcine Mendil, Florian Seka 1

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CONTENT

2

I. Objectives

II. Infrastructure solutions

III. Experiments

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OBJECTIVES

I-Objectives II- Infrastructure solutions III-Experiments

Browser

Bloc

_SIPphone

(4)

OBJECTIVES

Browser

Bloc

_SIPphone

WebRTC

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OBJECTIVES

Browser

Bloc

_SIPphone

WebRTC

(6)

WEBRTC

4 I-Objectives II- Infrastructure solutions III-Experiments

(7)

SIPML 5

Browser

SipML5

Bloc

SIPphone

(8)

SIPML 5

Browser

SipML5

Sip stack

WebRTC

Bloc

SIPphone

(9)

ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

SIPphone

(10)

ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

Registrar

Proxy SIP

RTP

Engine

HTTP

server

Websocket

server

SIPphone

(11)

ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

Registrar

Proxy SIP

RTP

Engine

HTTP

server

Websocket

server

SIPphone

HTML

webapp.js

HTTP GET

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ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

Registrar

Proxy SIP

RTP

Engine

HTTP

server

Websocket

server

SIPphone

HTML

webapp.js

HTTP GET SIP ov er W S SIP SIP

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ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

Registrar

Proxy SIP

RTP

Engine

HTTP

server

Websocket

server

SIPphone

HTML

webapp.js

HTTP GET SIP ov er W S SIP SIP SRTP RTP

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ARCHITECTURE

Browser

SipML5

Sip stack

WebRTC

Bloc

Registrar

Proxy SIP

RTP

Engine

HTTP

server

Websocket

server

SIPphone

HTML

webapp.js

HTTP GET SIP ov er W S SIP SIP SRTP RTP

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• Proxy and server SIP :

-

Asterisk v 11.2.2

-

Additional patch

for VP8 support

OUR SOLUTION

7

Asterisk

11.2.2

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SCENARIOS AND TESTS

8

sipML5

javascript

WebRTC

FireBug

PC

Virtual Machine

Asterisk

RTP debug

SIP debug

(CLI)

Wireshark

eth0 I-Objectives II- Infrastructure solutions III-Experiments

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SCENARIO 1: AUDIO CALL

• Scenario : an audio call between a

browser and a softphone

• Registration is performed

• Need a websocket server and a

proxy SIP (provided by Asterisk)

• VM network is on bridge

9

Chrome

192.168.0.45 192.168.0.46 192.168.0.11 Asterisk 192.168.0.25 g711

X Lite

g711 Host machine

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AUDIO CALL

CALL FLOW

Browser Softphone (already registered) Asterisk INVITE SDP 401 Unauthorized ACK INVITE SDP 100 Trying WS [INVITE SDP] 180 Ringing WS [200 OK] RTP WS [100 Trying] WS [180 Ringing] SRTP 200 OK

• Signaling encapsuled in

Websocket

Les trames SRTP ne sont pas encapsulées dans du websocket.

Notre version de

wireshark ne reconnait pas SRTP, il indique que c’est sur de l’UDP.

WS[REGISTER]

WS[401 Unauthorized]

WS[REGISTER]

10

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AUDIO CALL

CALL FLOW

• Signaling encapsuled in

Websocket

Notre version de

WS[REGISTER]

10

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AUDIO CALL

CALL FLOW

• Signaling encapsuled in

Websocket

Notre version de

WS[REGISTER]

10

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SCENARIO II:

AUDIOCONFERENCE

• adding modules in Asterisk:

MeetMe, ConfBridge

• Dial-In

• DTMF in SIP INFO

192.168.0.33 192.168.0.46 192.168.0.45 11 192.168.0.11 Asterisk Host machine 192.168.0.25 g711 g711 g711

LinPhone

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SCENARIO III:

PRESENCE

12 Browser Asterisk WS [NOTIFY] WS [200 OK] WS[SUSCRIBE] WS[401 Unauthorized] WS[200 OK] WS[SUSCRIBE] Status of userX ? I-Objectives II- Infrastructure solutions III-Experiments

Change of userX’s status WS [NOTIFY] WS [200 OK] 192.168.0.45 192.168.0.46 192.168.0.11 Asterisk 192.168.0.25 g711

X Lite

g711 Host machine

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SCENARIO III:

PRESENCE

12 Browser Asterisk WS [NOTIFY] WS [200 OK] WS[SUSCRIBE] WS[401 Unauthorized] WS[200 OK] WS[SUSCRIBE] Status of userX ? I-Objectives II- Infrastructure solutions III-Experiments

Change of userX’s status WS [NOTIFY] WS [200 OK] 192.168.0.45 192.168.0.46 192.168.0.11 Asterisk 192.168.0.25 g711

X Lite

g711 Host machine

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SCENARIO IV: VIDEO

• Works between softphones

using h264, h263, VP8

• Asterisk needs to be

patched to be

VP8-compliant

192.168.0.25 192.168.0.46 13 192.168.0.11 Asterisk 192.168.0.25 h.264 h.264

X Lite

iDoubs

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CONCLUSION

• WebRTC:

-

only VP8 available

-

works only with Chrome and Firefox

• Asterisk:

-

No video transcoding

‣

external transcoder: webrtc2sip ?

-

WebRTC users ≠ Softphone users

• OUR HTML5

CLIENT

16

• Deployed on

Asterisk HTTP

Server

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APPENDIX

• SIP messages encapsulated in WebSocket.

• No WebSocket on media plan.

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APPENDIX

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APPENDIX

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VARIABILITY OF TESTS

OS du PBX

CentOS

Ubuntu

PBX

PIAF-Green

Asterisk

Kamailo

OverSIP

OS

utilisateur

Windows 8

OS X

Ubuntu

Android

iOS

Softphone

SipInside,

_{X Lite}

Telephone,

_iDoubs

Zoiper

Sipdroid

_Media5-fone

Linphone,

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CONFERENCE

CALL FLOW

Computer _Asterisk Computer _Asterisk _Softphone

INVITE SDP 401 Unauthorized ACK INVITE SDP 100 Trying 200 OK ACK RTP WS[INVITE SDP] WS[401 Unauthorized] WS[ACK] WS[INVITE SDP] WS[100 Trying] WS[200 OK] WS[ACK] UDP WS[INVITE SDP] WS[401 Unauthorized] WS[ACK] WS[INVITE SDP] WS[100 Trying] WS[200 OK] WS[ACK] UDP