Signaling Workshop. Tsahi 11/25/2013

Transcription

1 1

2 Signaling Workshop Tsahi 2

3 Speakers Peter Dunkley Technical Director Crocodile RCS Ltd Kevin Wiseman Chief Architect CafeX Rod Apeldoorn EasyRTC Server Lead Priologic 3

4 Peter Dunkley Technical Director Crocodile RCS Ltd SIP OVER WEBSOCKETS 4

5 What is SIP over WebSockets It is exactly the same SIP as SIP over TCP, SIP over TLS, and SIP over UDP just over WebSockets instead It can be secure by using Secure WebSockets It is about to become an RFC currently in the IETF editors queue It is widely supported by many open-source projects 5

6 A quick comparison Standards Based SIP over WebSockets XMPP (BOSH/WebSockets) Proprietary Open-source support High Medium Low Vendor tie-in prevention High Medium Low Ease of use High Medium High* Client performance Medium Medium High* Network performance High Medium Medium* Ease of interoperability High Medium Low* Standalone applications High High High* Use existing media servers High Low Low * Dependent on the proprietary option chosen could be made better or worse depending on what you chose! 6

7 Open source support and vendor tiein prevention SIP over WebSockets At least four well tested open-source Javascript client stacks At least six well tested open-source server stacks XMPP (BOSH/WebSockets) At least two well tested open-source Javascript client stacks At least three open-source server stacks Proprietary Many options (even open-source options) but all different and incompatible - many are vendor specific 7

8 Ease of use SIP over WebSockets It is not hard you are not implementing the signalling yourself Many client library choices with different APIs and complexities Many SDK vendors working to make it even easier for you XMPP (BOSH/WebSockets) It is not hard you are not implementing the signalling yourself Limited client library choices mean that it is harder than it should be Proprietary Might well be very easy (but might not be) it depends on your technology choices No easier than SIP if you choose the right SIP client library 8

9 Client and network performance SIP over WebSockets Javascript client libraries can be large, but minification and app-cache use mitigate this almost completely The network can have very high performance while making use of years of developments in real-time signalling and scaling XMPP (BOSH/WebSockets) Javascript client libraries can be large, but minification and app-cache use mitigate this almost completely The network will be high performance as long as you do not require interoperability Proprietary The client library may be small (depending on your vendor choice) The network may be high performance (depending on your vendor choice) as you do not require interoperability 9

10 Interoperable and standalone The triangle vs the trapezoid All options are equally suitable for use with the triangle SIP over WebSockets is better for use with the trapezoid Avoid gateways whenever you can (they add network cost and complexity) Using SIP makes your application compatible with most existing media servers (commercial and open-source) 10

11 The Triangle Server UA Media UA 11

12 The Trapezoid Server Signalling Server UA Media UA 12

13 Use cases for SIP over WebSockets SIP is well suited for anything requiring interoperability Conferencing do you really want to exclude the guy travelling who can t get (or afford) a mobile data connection? Online education why shouldn t I be able to listen to lectures through other routes? Telemedicine a huge boon for people living in remote areas (aren t those the ones who struggle to get online?) Call centres can I afford to exclude customers who can t use (or don t want to use) WebRTC? Many of the applications that need to interoperate are high-value 13

14 Kamailio: WebSocket Handshake... tcp_accept_no_cl=yes... event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($hdr(upgrade)=~"websocket" && $hdr(connection)=~"upgrade" && $rm=~"get") { Cookie:) # Validate as required (Host:, Origin:, } if (ws_handle_handshake()) exit; } xhttp_reply("404", "Not Found", "", ""); 14

15 Kamailio: Request Handling modparam( nathelper registrar, received_avp, $avp(received) )... request_route { route(reqinit); route(wsdetect);... route[wsdetect] { if (proto == WS proto == WSS) { force_rport(); if (is_method( REGISTER )) { fix_nated_register(); } else if (is_method( INVITE NOTIFY SUBSCRIBE )) { add_contact_alias(); } } }... route[withindlg] { if (has_totag()) { if (loose_route()) { if (!isdsturiset()) { handle_ruri_alias(); }... 15

16 Kamailio: Response Handling onreply_route { if ((proto == WS proto == WSS) && status =~ [12][0-9][0-9] ) { add_contact_alias(); } } 16

17 Kamailio: RTP Profile Conversion (1) modparam( rtpproxy-ng, rtpproxy_sock, udp:localhost:22223 )... route[location] {... t_on_failure( UA_FAILURE ); }... failure_route[ua_failure] { if (t_check_status( 488 ) && sdp_content()) { if (sdp_get_line_startswith( $avp(mline), m= )) { if ($avp(mline) =~ SAVPF )) { $avp(rtpproxy_offer_flags) = froc-sp ; $avp(rtpproxy_answer_flags) = froc+sp ; } else { $avp(rtpproxy_offer_flags) = froc+sp ; $avp(rtpproxy_answer_flags) = froc-sp ; } # In a production system you probably need to catch # RTP/SAVP and RTP/AVPF and handle them correctly # too } append_branch(); rtpproxy_offer($avp(rtpproxy_offer_flags)); t_on_reply( RTPPROXY_REPLY ); route(relay); } }... 17

18 Kamailio: RTP Profile Conversion (2) modparam( rtpproxy-ng, rtpproxy_sock, udp:localhost:22223 )... failure_route[ua_failure] {... t_on_reply( RTPPROXY_REPLY ); route(relay); } onreply_route[rtpproxy_reply] { if (status =~ 18[03] ) { # mediaproxy-ng currently only supports SRTP/SDES early media # won't work so strip it out now to avoid problems change_reply_status(180, Ringing ); remove_body(); } else if (status =~ 2[0-9][0-9] && sdp_content()) { rtpproxy_answer($avp(rtpproxy_answer_flags)); } }... 18

19 Authentication (1) You do not need accounts on the SIP network You can federate with existing identity providers (Facebook, Google+, LinkedIn, Twitter, your own) There is a Kamailio module designed for this (and implementing it elsewhere is easy) 19

20 Authentication (2) Web Service OAuth2 Calling UA Shared secret communication link not required You don t have to create or manage accounts on the SIP Proxy/registrar SIP Proxy Called UA 20

21 Kamailio: Authentication (1)... tcp_accept_no_cl=yes... modparam( auth_ephemeral, secret, kamailio_rules )... modparam( htable, htable, wsconn=>size=8; )... event_route[xhttp:request] {... # URI format is /?username=foo&password=bar $var(uri_params) = $(hu{url.querystring}); $var(username) = $(var(uri_params){param.name,username,&}); $var(password) = $(var(uri_params){param.name,password,&}); # Note: username and password could also have been in a Cookie: header if (!autheph_authenticate( $var(username), $var(password) )) { xhttp_reply( 403, Forbidden,, ); exit; } if (ws_handle_handshake()) { $sht(wsconn=>$si:$sp::username) = $var(username) exit; }... event_route[websocket:closed] { $var(regex) = $si + : $sp +.* ; sht_rm_name_re( wsconn=>$var(regex) ); } 21

22 Kamailio: Authentication (2)... request_route { route(reqinit); route(wsdetect);... if (!(proto == WS proto == WSS)) route(auth);... route[wsdetect] { if (proto == WS proto == WSS) { $var(username) = (str) $sht(wsconn=>$si:$sp::username); if ($var(username) == $null $var(username) == ) { send_reply( 403, Forbidden ); ws_close(1008, Policy Violation ); exit; } if (!autheph_check_timestamp( $var(username) ) (is_method( REGISTER PUBLISH ) &&!autheph_check_to( $var(username) )) (!has_totag() &&!autheph_check_from( $var(username) ))) { send_reply( 403, Forbidden ); ws_close(1008, Policy Violation ); exit; } force_rport();... 22

23 Questions? Code:

24 Kevin Wiseman Chief Architect CaféX WHAT IS SIGNALLING 24

25 The Basics WebRTC Spec does not define the signalling It leaves that open to the implementer It does define the media descriptor exchange Utilises the Session Description Protocol 25

26 What is signalling Communications session control from one party to another party Typically via some location service E.g. SIP Registrar, social network, games service Describes the requests and responses Make call, end call, hold, resume, transfer etc Ringing, answer, rejected, established etc Mechanism for exchanging media description SDP offer/answer exchange 26

27 Session Description Protocol Used to negotiate media between parties Media (audio, video), Ports, Codecs, ICE etc offer/answer exchange The good news: Browser generates and consumes SDP offers & answers The bad news: WebRTC SDP places specific requirements on SDP that most existing telecom entities do not support 27

28 Signalling isn't the hard piece with WebRTC Media and media description are 28

29 Closed Systems Real time coms within Games, Social Networks etc No compelling reason to adopt any one approach over another Already know who is who and how to communicate No integration into existing telcoms system Extend existing control channel for SDP offer/answer, or Utilise 3 rd party services to provide real time coms 29

30 WebRTC Gateway Proprietary signalling over HTTP or WebSocket between browser and gateway Gateway makes sense if: The call scenarios you need to support are standard well defined UC features Voice & video: Make call, answer call, end call, transfer call, N-way call IM&P: Send message, receive message, set presence Application Event Distribution You want to embed UC features as part of a service Your SIP infrastructure has limited support for ICE/STUN and Multiplexing RTP etc This is the use case we are seeing almost exclusively Customers wanting to embed UC features as part of an overall service Challenges Dependency on gateway exposing features JSON call control over WebSocket Fusion Web Gateway SIP CaféX Implementation Client SDK SIP Network Browser Fusion Media Broker SRTP De-ICEd, de-stunned & de-multiplexed RTP 30

31 SIP over Websockets SIP over web sockets makes sense if: The call scenarios you need to support to the browser require all the features of SIP You have developers that can make full use of SIP features on the browser Your SIP end points already support ICE/STUN and Multiplexing RTP etc Challenges SIP stack/ua in the browser (dependency on large and complex JS in Browser) SIP interworking with yet another 3rd party SIP stack Security opening up all the features (and potential security holes) of SIP to the public internet is dangerous No benefit to most existing SIP systems still have to add web sockets support, still have to add SDP and Media translation SIP over Websocket Fusion Web Gateway SIP If Café X had used SIP over WebSockets JS SIP Stack JS SIP UA SIP Network Browser RTP Fusion Media Broker De-ICEd, de-stunned & de-multiplexed RTP 31

32 Web Dev & Signalling Most Web Devs don t know or even need to know the signalling protocol Busy enough already Web Devs expect a rich functional high level API Efficiency & ease of use Don t want to reinvent the wheel every time they want to add RTC to a web app If no API provided the first thing a web dev will do is write one and re-use next time they need to use the service 32

33 API Example Initialisation <script src=" //Get hold of the sessionid however your app needs to var sessionid = getmysessionid(); //Google provide a stun server which you can use or you can use your own. //Providing any empty array will result in no stun messages being sent. var stunservers=["stun.l.google.com:19302"]; //Set up initialization success callback before calling start UC.onInitialised = function() { //perform tasks associated with successful initialization such as registering listeners on UC objects }; //Set up initialization failure callback before calling start UC.onInitialisedFailed = function() { //perform tasks associated with initialization failure}; //Start my UC session using the Session ID and stun server list UC.start(sessionID, stunservers); 33

34 API Example Media Streams window.webkiturl.createobjecturl.uc.phone.onremotemediastream = function(remotemediastream) { //Configure the streams, this can be used to set up visibility of elements and //set the elements src to that of the remote stream, the remotemediastream must be //added to the page in order to receive audio or and video. video.src = window.webkiturl.createobjecturl(remotemediastream); }; UC.phone.onLocalMediaStream = function(localmediastream) { //As with the remote media stream, you should add the localmediastream to the page //in order to allow the framework to playback local (ie, webcam) feedback preview.src = window.webkiturl.createobjecturl(localmediastream); }; 34

35 API Example Make Call var call; //A method to call from the UI to make a call function makecall(addresstocall) { //Create a call object from the framework and save it somewhere call = UC.phone.createCall(addressToCall); //Set what to do when the remote party ends the call call.onended = function() { alert("call Ended"); }; //Set up what to do if the callee is busy, inform your user etc call.onbusy = function() { alert("the callee was busy"); }; }; //Dial the call call.dial(); //A method to call from the UI to end a current call function endcall() { call.end(); }; 35

36 Rod Apeldoorn EasyRTC Server Lead Priologic CUSTOM SIGNALING 36

37 Benefits Over SIP Faster on the client side Easier for developers to understand Less restrictive 37

38 Example Custom Message Types (from EasyRTC) WebRTC Core candidate offer answer reject Application Level authenticate hangup geticeconfig roomjoin roomdata setpresence filesoffer Many more 38

39 Why Combine WebRTC Signaling with Application Servers? Authentication Call logging Call control Combine with application logic Client connects to just one server Why SIP + Presence + Application servers? Will SIP Gateways offer JSON signaling? Yes! 39

40 Transports Websockets Available in all modern browsers Fast + Responsive + Securable Maintains open socket Servers have to deal with concurrent socket limits XHR Polling AKA HTTP Long Polling Easy + Securable To use: XMLHttpRequest API jquery.ajax() Used by Google AppRTC Demo 40

41 Transports JSONP + CORS The original popular method for DHTML Cross site scripting issues Cross-Origin Resource Sharing can be setup Still a valid fallback Especially for older browsers Other XMPP (Jabber) Instant messengers Local Bluetooth USB / Serial WebRTC Data Channels Example coming! 41

42 Cisco Jabber + EasyRTC Cisco DX650 chat with Cisco or Non-Cisco web user WebRTC Across Devices and Transports Built using Cisco Jabber SDK EasyRTC Opensource 42

43 Using a Websocket Library General Benefits Cross browser support Easy message sending Easy event handling Connect / Disconnect / Message Fallbacks to XHR or JSONP Why EasyRTC uses Socket.io Most popular for Node.js Client API s in many languages JavaScript / C++ / ObjC / Java Message Acknowledgments Why recode what s done well? 43

44 Private WebRTC Signaling 1. Connect users to servers via Websockets 2. Establish DataChannels between users on same servers 3. Establish WebRTC Peer Connection between User 1 and 3 Signals sent via DataChannel User 2 acts as a relay Neither server aware of final connection 44

45 Private WebRTC Signaling 1. Connect users to servers via Websockets 2. Establish DataChannels between users on same servers 3. Establish WebRTC Peer Connection between User 1 and 3 Signals sent via DataChannel User 2 acts as a relay Neither server aware of final connection 45

46 Private WebRTC Signaling 1. Connect users to servers via Websockets 2. Establish DataChannels between users on same servers 3. Establish WebRTC Peer Connection between User 1 and 3 Signals sent via DataChannel User 2 acts as a relay Neither server aware of final connection 46

47 Q&A 47