WEBRTC : EXPLORATION THROUGH THE QUESTION OF INTEROPERABILITY WITH SIP

Transcription

1 WEBRTC : EXPLORATION THROUGH THE QUESTION OF INTEROPERABILITY WITH SIP Soutenance 17/06/2013 Ornella Annicchiarico, Benoit Le Quéau, Mouhcine Mendil, Florian Seka 1

2 CONTENT I. Objectives II. Infrastructure solutions III. Experiments IV. Demonstration 2

3 OBJECTIVES Browser Bloc SIPphone 3

4 OBJECTIVES Browser Bloc SIPphone WebRTC 3

5 OBJECTIVES Browser Bloc SIPphone WebRTC 3

6 WEBRTC 4

7 Browser SIPML 5 Bloc SIPphone SipML5 5

8 Browser SIPML 5 Bloc SIPphone SipML5 Sip stack WebRTC 5

9 Browser ARCHITECTURE Bloc SipML5 Sip stack WebRTC SIPphone 6

10 Browser ARCHITECTURE HTTP server Bloc Websocket server SipML5 Sip stack Registrar Proxy SIP WebRTC RTP Engine SIPphone 6

11 Browser ARCHITECTURE HTML webapp.js HTTP GET HTTP server Bloc Websocket server SipML5 Registrar Proxy SIP Sip stack WebRTC RTP Engine SIPphone 6

12 Browser ARCHITECTURE HTML webapp.js HTTP GET HTTP server Bloc SipML5 SIP over WS Websocket server SIP Registrar Proxy SIP Sip stack WebRTC RTP Engine SIP SIPphone 6

13 Browser ARCHITECTURE HTML webapp.js HTTP GET HTTP server Bloc SipML5 SIP over WS Websocket server SIP Registrar Proxy SIP Sip stack WebRTC SRTP RTP Engine SIP SIPphone 6 RTP

14 Browser ARCHITECTURE HTML webapp.js HTTP GET HTTP server Bloc SipML5 SIP over WS Websocket server SIP Registrar Proxy SIP Sip stack WebRTC SRTP RTP Engine SIP SIPphone 6 RTP

15 OUR SOLUTION Proxy and server SIP : - Asterisk v Asterisk Additional patch for VP8 support 7

16 SCENARIOS AND TESTS sipml5 PC javascript WebRTC Virtual Machine eth0 Asterisk RTP debug SIP debug (CLI) FireBug Wireshark 8

17 SCENARIO 1: AUDIO CALL Scenario : an audio call between a browser and a softphone Host machine Asterisk Registration is performed Need a websocket server and a proxy SIP (provided by Asterisk) VM network is on bridge Chrome X Lite g711 g711 9

18 Browser WS[REGISTER] Asterisk Softphone (already registered) AUDIO CALL CALL FLOW WS[401 Unauthorized] WS[REGISTER] WS[200 OK] INVITE SDP 401 Unauthorized ACK INVITE SDP Signaling encapsuled in 100 Trying Websocket WS [INVITE SDP] WS [100 Trying] WS [180 Ringing] 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 [200 OK] SRTP 180 Ringing 200 OK RTP 10

19 Browser WS[REGISTER] Asterisk Softphone (already registered) AUDIO CALL CALL FLOW WS[401 Unauthorized] WS[REGISTER] WS[200 OK] INVITE SDP 401 Unauthorized ACK INVITE SDP Signaling encapsuled in 100 Trying Websocket WS [INVITE SDP] WS [100 Trying] WS [180 Ringing] 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 [200 OK] SRTP 180 Ringing 200 OK RTP 10

20 Browser WS[REGISTER] Asterisk Softphone (already registered) AUDIO CALL CALL FLOW WS[401 Unauthorized] WS[REGISTER] WS[200 OK] INVITE SDP 401 Unauthorized ACK INVITE SDP Signaling encapsuled in 100 Trying Websocket WS [INVITE SDP] WS [100 Trying] WS [180 Ringing] 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 [200 OK] SRTP 180 Ringing 200 OK RTP 10

21 SCENARIO II: AUDIOCONFERENCE Host machine Asterisk adding modules in Asterisk: MeetMe, ConfBridge LinPhone Dial-In DTMF in SIP INFO g g g711 11

22 Browser Asterisk SCENARIO III: PRESENCE Status of userx? WS[SUSCRIBE] WS[401 Unauthorized] Host machine Asterisk WS[SUSCRIBE] WS[200 OK] WS [NOTIFY] WS [200 OK] WS [NOTIFY] Change of userx s status X Lite WS [200 OK] g711 g711 12

23 Browser Asterisk SCENARIO III: PRESENCE Status of userx? WS[SUSCRIBE] WS[401 Unauthorized] Host machine Asterisk WS[SUSCRIBE] WS[200 OK] WS [NOTIFY] WS [200 OK] WS [NOTIFY] Change of userx s status X Lite WS [200 OK] g711 g711 12

24 SCENARIO IV: VIDEO Host machine Asterisk Works between softphones using h264, h263, VP Asterisk needs to be patched to be VP8- compliant X Lite idoubs h.264 h

25 CONCLUSION WebRTC: - only VP8 available - works only with Chrome and Firefox Asterisk: - No video transcoding external transcoder: webrtc2sip? - WebRTC users Softphone users other solutions: jssip/oversip... 14

26 DEMONSTRATION!

27 OUR HTML5 CLIENT Deployed on Asterisk HTTP Server 16

28 APPENDIX SIP messages encapsulated in WebSocket. No WebSocket on media plan.

29 APPENDIX

30 APPENDIX

31 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 Linphone, Media5-fone Navigateur Firefox Nightly Chrome Bowser

32 CONFERENCE CALL FLOW Computer Asterisk Computer Asterisk Softphone WS[INVITE SDP] WS[INVITE SDP] INVITE SDP WS[401 Unauthorized] WS[ACK] WS[INVITE SDP] WS[100 Trying] WS[200 OK] WS[ACK] WS[401 Unauthorized] WS[ACK] WS[INVITE SDP] WS[100 Trying] WS[200 OK] WS[ACK] 401 Unauthorized ACK INVITE SDP 100 Trying 200 OK ACK UDP UDP RTP