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Signaling Workshop

Tsahi Levent-Levitsahil@bloggeek.me@tsahil

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Speakers• Peter Dunkley– Technical Director– Crocodile RCS Ltd

• Kevin Wiseman – Chief Architect– CafeX

• Rod Apeldoorn – EasyRTC Server Lead– Priologic

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SIP OVER WEBSOCKETS

Peter DunkleyTechnical DirectorCrocodile RCS Ltd

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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

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A quick comparisonStandards Based

ProprietarySIP over WebSockets XMPP (BOSH/WebSockets)

Open-source support High Medium LowVendor tie-in prevention High Medium LowEase 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!

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Open source support and vendor tie-in 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

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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

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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

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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)

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The Triangle

UA UAMedia

Server

SignallingSignalling

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The Trapezoid

UA UA

Server Signalling

Media

Server

SignallingSigna

lling

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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

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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") {

# Validate as required (Host:, Origin:, Cookie:)

if (ws_handle_handshake())exit;

}

xhttp_reply("404", "Not Found", "", "");}

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Kamailio: Request Handlingmodparam(“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();

}...

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Kamailio: Response Handling

onreply_route {if ((proto == WS || proto == WSS)

&& status =~ “[12][0-9][0-9]”) {add_contact_alias();

}}

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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);

}}...

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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 problemschange_reply_status(180, “Ringing”);remove_body();

} else if (status =~ “2[0-9][0-9]” && sdp_content()) {rtpproxy_answer($avp(rtpproxy_answer_flags));

}}...

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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)

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Authentication (2)

Web Service

SIP Proxy

Calling UA

Called UA

SIP

SIP

RESTOAuth2

Shared secret – communication

link not required

You don’t have to create or manage accounts on the SIP Proxy/registrar

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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)”);

}

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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();...

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Questions?

Code: https://github.com/crocodilertc

Email: peter.dunkley@crocodilertc.net

Twitter: @pdunkley

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WHAT IS SIGNALLING

Kevin WisemanChief ArchitectCaféX

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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

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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

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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

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Signalling isn't the hard piece with WebRTC

Media and media description are

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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 3rd party services to provide real time coms

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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

Fusion Web

Gateway

Fusion Media Broker

Browser

SIP Network

JSON call control over WebSocket

SIP

SRTP

De-ICEd, de-STUNned &de-multiplexedRTP

Client SDK

CaféX Implementation

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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

Fusion Web

Gateway

Fusion Media Broker

Browser

SIP Network

SIP over Websocket

SIP

RTP

De-ICEd, de-STUNned &de-multiplexedRTP

JS SIPStack

JS SIP UA

If Café X had used SIP over WebSockets

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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

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API Example – Initialisation<script src="https://<gateway_address>:<8080>/gateway/fusion-client-sdk.js"></script>

//Get hold of the sessionID however your app needs tovar 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 startUC.onInitialised = function() {

//perform tasks associated with successful initialization such as registering listeners on UC objects};

//Set up initialization failure callback before calling startUC.onInitialisedFailed = function() {

//perform tasks associated with initialization failure};

//Start my UC session using the Session ID and stun server listUC.start(sessionID, stunServers);

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API Example – Media Streamswindow.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) feedbackpreview.src = window.webkitURL.createObjectURL(localMediaStream);

};

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API Example – Make Callvar call;

//A method to call from the UI to make a callfunction 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 callcall.onEnded = function() { alert("Call Ended"); };

//Set up what to do if the callee is busy, inform your user etccall.onBusy = function() { alert("The callee was busy"); };

//Dial the callcall.dial();

};

//A method to call from the UI to end a current callfunction endCall() { call.end(); };

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CUSTOM SIGNALING

Rod Apeldoorn EasyRTC Server LeadPriologic

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

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Example Custom Message Types(from EasyRTC)

WebRTC Core• candidate• offer• answer• reject

Application Level• authenticate• hangup• getIceConfig• roomJoin• roomData• setPresence• filesOffer• Many more…

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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!

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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

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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!

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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

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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?

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Private WebRTC Signaling1. Connect users to

servers via Websockets2. 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

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Private WebRTC Signaling1. Connect users to

servers via Websockets2. 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

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Private WebRTC Signaling1. Connect users to

servers via Websockets2. 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

http://bit.ly/1iq6v8Dhttp://bit.ly/1iq6v8D

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Q&A