jssip - npm Package Compare versions

Comparing version 3.9.4 to 3.10.0

CHANGELOG.md

 CHANGELOG
 =========
 
+### 3.10.0
+
+* Fix typescript typings (#648).
+* New UA config param 'extra_headers' (#774).
+
+
 ### 3.9.4
 
-
 * Dialog/RequestSender: Avoid circular dependency (#788).
@@ -11,6 +16,4 @@ * Fixed REFER NOTIFY parsing in accordance to RFC 3515 section 2.4.5 (#767).
 
-
 ### 3.9.3
 
-
 * UA: Add TS types for 'newOptions' event (closes #777).
@@ -21,3 +24,2 @@
 
-
 * Dialog/RequestSender: Fix reference to undefined method (#787).
@@ -28,3 +30,2 @@
 
-
 * Fix RTCSession 'confirmed' event type description (#770). Thanks @stefang42.
@@ -35,3 +36,2 @@
 
-
 * Enable custom From-Header for outbound Message (#753). Thanks @gebsl.
@@ -42,3 +42,2 @@
 
-
 * RTCSession: Fix for overlapping invites while refreshing session (PR #734). Credits to @makstheimba.
@@ -50,3 +49,2 @@
 
-
 * RTCSession: allow SDP renegotiation before ICE gathering is completed. Credits to @markusatm.
@@ -57,3 +55,2 @@
 
-
 * Add support for sending and responding to SIP OPTIONS. Credits to @simonlindberg.
@@ -64,3 +61,2 @@
 
-
 * ReferSubscriber: fix, case sensitivity.
@@ -71,3 +67,2 @@
 
-
 * RTCSession: allow overriding the Referred-By header on REFER. Credits to @jp1987.
@@ -78,3 +73,2 @@
 
-
 * RTCSession: allow sending DTMFs if 1XX was received. Credits to @BlindChickens.
@@ -85,3 +79,2 @@
 
-
 * Use a proper Logger class (fix from previous release)
@@ -92,3 +85,2 @@
 
-
 * Use a proper Logger class.
@@ -99,3 +91,2 @@
 
-
 * Registrator: Enhance Contact matching in Register 2XX reponse. Credits to @stefang42.
@@ -106,3 +97,2 @@
 
-
 * WebSocketInterface: Fix values of UA.disconnected event (#701). Credits to @kkozlik.
@@ -113,3 +103,2 @@
 
-
 * Support 2 bytes long UTF-8 chars in the SDP (#693). Credits to [e-mail].
@@ -120,3 +109,2 @@
 
-
 * RTCSession: close media stream even if session is in terminated (#683). Credits to @kkozlik.
@@ -127,3 +115,2 @@
 
-
 * Enable custom From-Header for outbound calls (#677). Thanks @fabrykowski.
@@ -134,3 +121,2 @@
 
-
 * Make authorization_jwt writable (#676). Thanks @fabrykowski.
@@ -141,3 +127,2 @@
 
-
 * Add missing typescript definitions. Thanks @fabrykowski.
@@ -148,3 +133,2 @@
 
-
 * RTCSession: Properly use RTCDtmf sender enqueue (#674). Thanks @ikq for reportig and testing.
@@ -155,3 +139,2 @@
 
-
 * UA configuration option 'session_timers_force_refresher' (#671). Credits to @kkozlik.
@@ -162,3 +145,2 @@
 
-
 * RTCsession: Fix, missing progress event for empty provisional responses. Credits to @stefang42.
@@ -169,3 +151,2 @@
 
-
 * RTCSession: Fix, in dialog request processing error does not take the seesion into a failed state. Thanks @stefang42 for reportig.
@@ -175,3 +156,2 @@
 
-
 * RTCSession: Fix, fire progress after setRemoteDescription. Credits to @wangduanduan.
@@ -181,3 +161,2 @@
 
-
 * RTCSession: handle incoming BYE in WAITING_FOR_ANSWER state. #665. Credits to @karstenluedtke.
@@ -187,3 +166,2 @@
 
-
 * Registrator: improve expires maths. Credits to Roman Shpount.
@@ -193,3 +171,2 @@
 
-
 * Registrator: improve expires maths. Credits to Roman Shpount.
@@ -199,3 +176,2 @@
 
-
 * JsSIP.d.ts: import and export in two steps. #651. Credits to @ashlanderDesign.
@@ -205,3 +181,2 @@
 
-
 * Fix request cseq upon 491 response (#653). Credits to @mattdimeo.
@@ -212,3 +187,2 @@ * UA: Allow to change authorization_user (#652). Credits to @jose-lopes.
 
-
 * RTCSession: force gather candidates when iceRestart is true (#641). Credits to @jose-lopes.
@@ -219,3 +193,2 @@ * Fix crash if the request does not have Content-Type header (#650). Credits to @hxl-dy
 
-
 * Fix Via 'port' parser (#642).
@@ -225,3 +198,2 @@
 
-
 * Export Socket interface (#637).
@@ -231,3 +203,2 @@
 
-
 * Typescript typings (#627).
@@ -237,3 +208,2 @@
 
-
 * Clone optional objects (#625). Credits to @acharlop.
@@ -244,3 +214,2 @@ * RTCSession: fix, allow initial INVITE no SDP/offer (#624). Credits to @mgoodenUK.
 
-
 * Transactions: destroy transaction in Timer I after terminating. Thanks Juha Heinanen for reporting the issue.
@@ -250,3 +219,2 @@
 
-
 * RTCSession: disable remote hold state when receiving an INVITE without SDP (#613). Credits to @RobyMcAndrew.
@@ -257,3 +225,2 @@
 
-
 * RequestSender: fix Authorization header addition for jwt use.
@@ -264,3 +231,2 @@
 
-
 * Add `authorization_jwt` configuration parameter (#610). Credits to @voicenter.
@@ -271,3 +237,2 @@
 
-
 * RTCSession: don't relay on 'icecandidate' event with null candidate (#598). Thanks @skanizaj.
@@ -278,3 +243,2 @@
 
-
 * RTCSession: honor BYE while in WAITING_FOR_ACK state (#597). Thanks @Egorikhin.
@@ -285,3 +249,2 @@
 
-
 * Added NOTIFY to allowed methods (#593). Credits to @ikq.
@@ -292,3 +255,2 @@
 
-
 * Move connection recovery defaults to Constants (#593). Credits to @KraftyKraft.
@@ -299,3 +261,2 @@
 
-
 * Add referred-by header to refer messages (#572). Credits to @swysor.
@@ -306,3 +267,2 @@
 
-
 * Fix NameAddrHeader `display_name` handling (#573). Credits to @nicketson.
@@ -313,3 +273,2 @@
 
-
 * Add `.babelrc` into `.npmignore` (related to #489).
@@ -321,3 +280,2 @@ * Update deps.
 
-
 * Add debugging logs in DigestAuthentication.js (related to #561).
@@ -329,3 +287,2 @@ * Update deps.
 
-
 * Registrator: Don't check Contact header if final response is not 2XX (#558). Thanks @ikq for reporting.
@@ -337,3 +294,2 @@ * Update deps.
 
-
 * Registrator. Support multiple entries in the same Contact header field (#544).
@@ -344,3 +300,2 @@
 
-
 * RTCSession: fire 'sdp' event on renegotiation (#543).
@@ -351,3 +306,2 @@
 
-
 * UA: new 'sipEvent' event for out of dialog NOTIFY requests.
@@ -358,3 +312,2 @@
 
-
 * InviteClientTransaction: Add full route set to ACK and CANCEL requests. Thanks @nicketson.
@@ -366,3 +319,2 @@ * RTCSession: switch to tracks from deprecated stream API. Thanks @nicketson.
 
-
 * Fix typos thanks to the [LGTM](https://lgtm.com/projects/g/versatica/JsSIP/alerts/?mode=list) project.
@@ -374,3 +326,2 @@ * Update deps.
 
-
 * Remove `webrtc-adapter` dependency. It's up to the application developer whether to include it into his application or not.
@@ -382,3 +333,2 @@ * Update dependencies.
 
-
 * Revert previous release. Requires a mayor version upgrade for such a cosmetic change.
@@ -389,3 +339,2 @@
 
-
 * Close #521, #534. RTCSession: Fix 'connection' event order on outgoing calls.
@@ -396,3 +345,2 @@
 
-
 * Update deps.
@@ -404,3 +352,2 @@ * Add missing `error` in 'getusermediafailed' event (thanks @jonastelzio).
 
-
 * Close #519. Parser: Do not overwrite unknwon header fields. Thanks @rprinz08.
@@ -411,3 +358,2 @@
 
-
 * Include the NPM **events** dependency for those who don't use **browserify** but **webpack**.
@@ -418,3 +364,2 @@
 
-
 * RTCSession: Add Contact header to REFER request. Thanks Julien Royer for reporting.
@@ -425,3 +370,2 @@
 
-
 * Fix #511. Add missing payload on 'UA:disconnected' event.
@@ -432,3 +376,2 @@
 
-
 * Fix regression (#509): ua.call() not working if stream is given.
@@ -439,3 +382,2 @@
 
-
 * RTCSession: custom local description trigger support
@@ -446,3 +388,2 @@
 
-
 * RTCSession: prefer promises over callbacks for readability.
@@ -453,3 +394,2 @@
 
-
 * Config: #494. Switch Socket check order. Thanks 'Igor Kolosov'.
@@ -460,3 +400,2 @@
 
-
 * RTCSession: Fix #492. Add missing log line for RTCPeerConnection error.
@@ -467,3 +406,2 @@
 
-
 * Remove wrong NPM dependencies.
@@ -474,3 +412,2 @@
 
-
 * Fix parsing of NOTIFY bodies during a REFER transaction (fixes #493).
@@ -481,3 +418,2 @@
 
-
 * Config: new configuration parameter 'user_agent'
@@ -490,3 +426,2 @@ * RTCSession/Info: Fix. Call session.sendRequest() with the correct parameters
 
-
 * Fix #482 and cleanup Registrator.js
@@ -497,3 +432,2 @@
 
-
 * Produce ES5 tree and expose it as main in package.json (related to #472)
@@ -505,3 +439,2 @@ * Fix #481. ReferSubscriber: properly access RTCSession non-public attributes
 
-
 * RTCSession: emit 'sdp' event before creating offer/answer
@@ -512,3 +445,2 @@
 
-
 * DigestAuthentication: fix 'auth-int' qop authentication
@@ -520,3 +452,2 @@ * DigestAuthentication: add tests
 
-
 * New UA configuration parameter 'session_timers_refresh_method'. Thanks @michelepra
@@ -527,3 +458,2 @@
 
-
 * Fix improper call to userMediaSucceeded. Thanks @iclems
@@ -534,3 +464,2 @@
 
-
 * Registrator: add missing getter. Thanks Martin Ekblom.
@@ -541,3 +470,2 @@
 
-
 * Fix #473. Typo. Thanks @ikq.
@@ -548,3 +476,2 @@
 
-
 * Use promise chaining to prevent PeerConnection state race conditions. Thanks @davies147
@@ -555,3 +482,2 @@
 
-
 * Fix #421. Fire RTCSession 'peerconnection' event as soon as its created
@@ -562,3 +488,2 @@
 
-
 * Fix typo. Thanks @michelepra.
@@ -569,3 +494,2 @@
 
-
 * Tests: enable test-UA-no-WebRTC tests.
@@ -578,3 +502,2 @@ * WebSocketInterface: uppercase the via_transport attribute.
 
-
 * WebSocketInterface: Add 'via_transport' setter.
@@ -585,3 +508,2 @@
 
-
 * Fix typo on ES6 transpiling.
@@ -592,3 +514,2 @@
 
-
 * ES6 transpiling. Modernize full JsSIP code.
@@ -599,3 +520,2 @@
 
-
 * Dialog: ACK to initial INVITE could have lower CSeq than current remote_cseq.
@@ -606,3 +526,2 @@
 
-
 * RTCSession: process INFO in early state.
@@ -613,3 +532,2 @@
 
-
 * Fix #457. Properly retrieve ReferSubscriber. Thanks @btaens.
@@ -620,3 +538,2 @@
 
-
 * Fix #457. Support NOTIFY requests to REFER subscriptions without Event id parameter.
@@ -627,3 +544,2 @@
 
-
 * Update dependencies.
@@ -634,3 +550,2 @@
 
-
 * `Registrator`: Don't send a Register request if another is on progress. Thanks to Paul Grebenc.
@@ -641,3 +556,2 @@
 
-
 * `UA`: Add `registrationExpiring` event (#442). Credits to @danjenkins.
@@ -648,3 +562,2 @@
 
-
 * `RTCSession`: Emit "peerconnection" also for incoming calls.
@@ -655,3 +568,2 @@
 
-
 * Emit SDP before new `RTCSessionDescription`. Thanks to @StarLeafRob.
@@ -662,3 +574,2 @@
 
-
 * Generic SIP INFO support.
@@ -669,3 +580,2 @@
 
-
 * Fix #431. Fix UA's `disconnect` event by properly providing an object with all the documente fields (thanks @nicketson for reporting it).
@@ -676,3 +586,2 @@
 
-
 * Fix #428. Don't use `pranswer` for early media. Instead create an `answer` and do a workaround when the 200 arrives.
@@ -683,3 +592,2 @@
 
-
 * Update deps.
@@ -691,3 +599,2 @@ * Add more debug logs into `RTCSession` class.
 
-
 * Update deps.
@@ -699,3 +606,2 @@ * If ICE fails, terminate the session with status code 408.
 
-
 * Fix #426. Properly emit DTMF events.
@@ -706,3 +612,2 @@
 
-
 * Fix #418. Incorrect socket status on failure.
@@ -713,3 +618,2 @@
 
-
 * Close #419. Allow sending the DTMF 'R' key. Used to report a hook flash.
@@ -720,3 +624,2 @@
 
-
 * Remove `rtcninja` dependency. Instead use `webrtc-adapter`.
@@ -732,3 +635,2 @@ * `RTCSession:`: Remove `RTCPeerConnection` event wrappers. The app can access them via `session.connection`.
 
-
 * Improve library logs.
@@ -739,3 +641,2 @@
 
-
 * Update dependencies.
@@ -746,3 +647,2 @@
 
-
 * Fix #400. Corrupt NPM packege.
@@ -753,3 +653,2 @@
 
-
 * Fix #385. No CANCEL request sent for authenticated requests.
@@ -760,3 +659,2 @@
 
-
 * Fix `gulp-header` dependency version.
@@ -767,3 +665,2 @@
 
-
 * Export `JsSIP.WebSocketInterface`.
@@ -774,3 +671,2 @@
 
-
 * New 'contact_uri' configuration parameter.
@@ -786,3 +682,2 @@ * Remove Node websocket dependency.
 
-
 * Update dependencies.
@@ -793,3 +688,2 @@
 
-
 * `RTCSession`: new event on('sdp') to allow SDP modifications.
@@ -800,3 +694,2 @@
 
-
 * `RTCSession`: Allow multiple calls to `refer()` at the same time.
@@ -807,3 +700,2 @@
 
-
 * `UA`: `set()` allows changing user's display name.
@@ -815,3 +707,2 @@ * Ignore SDP answer in received ACK retransmissions (fix [367](https://github.com/versatica/JsSIP/issues/367)).
 
-
 * `RTCSession`: Also emit `peerconnection` event for incoming INVITE without SDP.
@@ -822,3 +713,2 @@
 
-
 * `RTCSession/ReferSubscriber`: Fix typo that breaks exposed API.
@@ -829,3 +719,2 @@
 
-
 * `RTCSession`: Make `refer()` method to return the corresponding instance of `ReferSubscriber` so the app can set and manage as many events as desired on it.
@@ -836,3 +725,2 @@
 
-
 * Add INFO method to allowed methods list
@@ -844,3 +732,2 @@ * Add SIP Code 424 RFC 6442
 
-
 * Apply changes of 0.7.16 also to browserified files under `dist/` folder.
@@ -851,3 +738,2 @@
 
-
 * Fix [337](https://github.com/versatica/JsSIP/issues/337). Consistenly indicate registration status through events.
@@ -858,3 +744,2 @@
 
-
 * Emit UA 'connected' event before sending REGISTER on transport connection
@@ -866,3 +751,2 @@ * Fix [355](https://github.com/versatica/JsSIP/pull/355 ). call to non existent `parsed.error` function. Thanks Stéphane Alnet @shimaore
 
-
 * Fix sips URI scheme parsing rule.
@@ -873,3 +757,2 @@
 
-
 * Fix. Don't lowercase URI parameter values. Thanks to Alexandr Dubovikov @adubovikov
@@ -880,3 +763,2 @@
 
-
 * Accept new `UA` configuration parameters `ha1` and `realm` to avoid plain SIP password handling ([issue 353](https://github.com/versatica/JsSIP/issues/353)).
@@ -888,3 +770,2 @@ * New `UA.set()` and `UA.get()` methods to set and retrieve computed configuration parameters in runtime.
 
-
 * Fix typo ("iceconnetionstatechange" => "iceconnectionstatechange"). Thanks to Vertika Srivastava.
@@ -895,3 +776,2 @@
 
-
 * Make `gulp` run on Node 4.0.X and 5.0.X.
@@ -902,3 +782,2 @@
 
-
 * `UA`: Add `set(parameter, value)` method to change a configuration setting in runtime (currently just "password" is implemented).
@@ -909,3 +788,2 @@
 
-
 * `RTCSession`: Add `resetLocalMedia()` method to reset the session local MediaStream by enabling both its audio and video tracks (unless the remote peer is on hold).
@@ -916,3 +794,2 @@
 
-
 * `RTCSession`: Add "sending" event to outgoing, a good chance for the app to mangle the INVITE or its SDP offer.
@@ -923,3 +800,2 @@
 
-
 * Update dependencies.
@@ -931,3 +807,2 @@ * Improve gulpfile.js.
 
-
 * Don't ask for `getUserMedia` in `RTCSession.answer()` if no `mediaConstraints` are provided.
@@ -938,3 +813,2 @@
 
-
 * Allow rejecting an in-dialog INVITE or UPDATE message.
@@ -945,3 +819,2 @@
 
-
 * FIX properly restart UA if start() is called while closing.
@@ -952,3 +825,2 @@
 
-
 * Update dependencies.
@@ -959,3 +831,2 @@
 
-
 * Update dependencies.
@@ -966,3 +837,2 @@
 
-
 * Add REFER support.
@@ -973,3 +843,2 @@
 
-
 * Don't keep URI params&headers in the registrar server URI.
@@ -981,3 +850,2 @@ * `RTCSession` emits `peerconnection` for outgoing calls once the `RTCPeerConnection` is created and before the SDP offer is generated (good chance to create a `RTCDataChannel` without requiring renegotiation).
 
-
 * Add callback to `update` and `reinvite` events.
@@ -988,3 +856,2 @@
 
-
 * Added a parser for Reason header.
@@ -995,3 +862,2 @@
 
-
 * Fix array iteration in `URI#toString()` to avoid Array prototype mangling by devil libraries such as Ember.
@@ -1002,3 +868,2 @@
 
-
 * Auto-register on transport connection before emitting the event.
@@ -1009,3 +874,2 @@
 
-
 * Update "rtcninja" dependencie.
@@ -1016,3 +880,2 @@
 
-
 * Don't terminate SIP dialog if processing of 183 with SDP fails.
@@ -1024,3 +887,2 @@ * Update dependencies.
 
-
 * Update "rtcninja" dependency.
@@ -1031,3 +893,2 @@
 
-
 * Update "rtcninja" dependency.
@@ -1038,3 +899,2 @@
 
-
 * RTCSession: Fix Invite Server transaction destruction.
@@ -1045,3 +905,2 @@
 
-
 * RTCSession: Handle session timers before emitting "accepted".
@@ -1053,3 +912,2 @@ * Fix issue with latest version of browserify.
 
-
 * Fix double "disconnected" event in some cases.
@@ -1060,3 +918,2 @@
 
-
 * Don't iterate arrays with (for...in) to avoid problems with evil JS libraries that add stuff into the Array prototype.
@@ -1067,3 +924,2 @@
 
-
 * Be more flexible receiving DTMF INFO bodies.
@@ -1074,3 +930,2 @@
 
-
 * Update dependencies.
@@ -1081,3 +936,2 @@
 
-
 * Terminate the call with a proper BYE/CANCEL/408/500 if request timeout, transport error or dialog error happens.
@@ -1089,3 +943,2 @@ * Fix "rtcninja" dependency problem.
 
-
 * `RTCSession`: Improve `isReadyToReOffer()`.
@@ -1096,3 +949,2 @@
 
-
 * `RTCSession`: Avoid calling hold()/unhold/renegotiate() if an outgoing renegotiation is not yet finished (return false).
@@ -1105,3 +957,2 @@ * `RTCSession`: Add `options` and `done` arguments to hold()/unhold/renegotiate().
 
-
 * `RTCSession:` Emit `iceconnetionstatechange` event.
@@ -1113,3 +964,2 @@ * Update "rtcninja" dependency to 0.4.0.
 
-
 * `RTCSession:` Include initially given `rtcOfferConstraints` in `sendReinvite()` and `sendUpdate()`.
@@ -1120,3 +970,2 @@
 
-
 * Properly keep mute local audio/video if remote is on hold, and keep it even if we re-offer. Also fix SDP direction attributes in re-offers according to current local and remote "hold" status.
@@ -1127,3 +976,2 @@
 
-
 * Update "rtcninja" dependency to 0.3.3.
@@ -1134,3 +982,2 @@
 
-
 * Fix "Session-Expires" default value to 90 seconds.
@@ -1141,3 +988,2 @@
 
-
 * Update "rtcninja" dependency to 0.3.2.
@@ -1148,3 +994,2 @@
 
-
 * Don't reply 405 "Method Not Supported" to re-INVITE even if the UA's "newRTCSession" event is not set.
@@ -1156,3 +1001,2 @@ * `RTCSession`: Allow extraHeaders in `renegotiate()`.
 
-
 * `RTCSession`: Don't ask for `getUserMedia()` in outgoing calls if `mediaConstraints` is `{audio:false, video:false}`. It is user's responsability to, in that case, provide `offerToReceiveAudio/Video` in `rtcOfferConstraints`.
@@ -1163,3 +1007,2 @@
 
-
 * ' UA.call()': Return the `RTCSession` instance.
@@ -1171,3 +1014,2 @@ * ' UA.sendMessage()': Return the `Message` instance.
 
-
 * `RTCSession`: Don't process SDPs in retranmissions of 200 OK during reINVITE/UDATE.
@@ -1180,3 +1022,2 @@ * `RTCSession`: Emit 'reinvite' when a reINVITE is received.
 
-
 * `RTCSession`: Don't override `this.data` on `answer()` (unless `options.data` is given).
@@ -1187,3 +1028,2 @@
 
-
 * `RTCSession#connect()`: Add `rtcAnswerContraints` options for later incoming reINVITE or UPDATE with SDP offer.
@@ -1197,3 +1037,2 @@ * `RTCSession#answer()`: Add `rtcOfferConstraints` options for later incoming reINVITE without SDP offer.
 
-
 * Bug fix. Properly cancel when only '100 trying' has been received.
@@ -1204,3 +1043,2 @@
 
-
 * Bug fix: Do not set "Content-Type: application/sdp" in body-less UPDATE requests.
@@ -1211,3 +1049,2 @@
 
-
 * Support for [Session Timers](https://tools.ietf.org/html/rfc4028).
@@ -1218,3 +1055,2 @@
 
-
 * [debug](https://github.com/visionmedia/debug) module.
@@ -1229,3 +1065,2 @@ * [rtcninja](https://github.com/ibc/rtcninja.js) module.
 
-
 * JsSIP runs in Node!
@@ -1236,3 +1071,2 @@ * The internal design of JsSIP has also been modified, becoming a real Node project in which the "browser version" (`jssip-0.5.0.js` or `jssip-0.5.0.min.js`) is generated with [browserify](http://browserify.org). This also means that the browser version can be loaded with AMD or CommonJS loaders.
 
-
 * [(3b1ee11)](https://github.com/versatica/JsSIP/commit/3b1ee11) Fix references to 'this'.
@@ -1243,3 +1077,2 @@
 
-
 * [(ca7702e)](https://github.com/versatica/JsSIP/commit/ca7702e) Fix #257. RTCMediaHandler: fire onIceCompleted() on next tick to avoid events race conditions in Firefox 33.
@@ -1249,3 +1082,2 @@
 
-
 * This version is included into the [Bower](https://bower.io/) registry which means `$ bower install jssip`.
@@ -1256,3 +1088,2 @@
 
-
 * (https://jssip.net/documentation/0.4.x/api/session) Hold/Unhold implementation
@@ -1295,3 +1126,2 @@ * (https://jssip.net/documentation/0.4.x/api/session) Mute/Unmute implementation
 ### 0.3.0
--------------------------------
 
@@ -1376,3 +1206,2 @@ * [(fea1326)](https://github.com/versatica/JsSIP/commit/fea1326) Don't validate configuration.password against SIP URI password BNF grammar (fix #74).
 
-
 * [(24e32c0)](https://github.com/versatica/JsSIP/commit/24e32c0d16ff5fcefd2319fc445a59d6fc2bcb59) UA configuration `password` parameter is now optional.
@@ -1391,3 +1220,2 @@ * [(ffe7af6)](https://github.com/versatica/JsSIP/commit/ffe7af6276915695af9fd00db281af51fec2714f) Bug fix: UA configuration `display_name` parameter.
 
-
 * First stable release with full website and documentation.
@@ -1399,4 +1227,3 @@ * Refactored sessions, message and events API.
 
-
 * First release. No documentation.
 

lib-es5/Config.js

@@ -50,2 +50,4 @@ "use strict";
   connection_recovery_min_interval: JsSIP_C.CONNECTION_RECOVERY_MIN_INTERVAL,
+  // Global extra headers, to be added to every request and response
+  extra_headers: null,
 
@@ -246,2 +248,28 @@ /*
       }
+    },
+    extra_headers: function extra_headers(_extra_headers) {
+      var _extraHeaders = [];
+
+      if (Array.isArray(_extra_headers) && _extra_headers.length) {
+        var _iterator2 = _createForOfIteratorHelper(_extra_headers),
+            _step2;
+
+        try {
+          for (_iterator2.s(); !(_step2 = _iterator2.n()).done;) {
+            var header = _step2.value;
+
+            if (typeof header === 'string') {
+              _extraHeaders.push(header);
+            }
+          }
+        } catch (err) {
+          _iterator2.e(err);
+        } finally {
+          _iterator2.f();
+        }
+      } else {
+        return;
+      }
+
+      return _extraHeaders;
     }
@@ -248,0 +276,0 @@ }

lib-es5/SIPMessage.js

@@ -70,5 +70,10 @@ "use strict";
     this.body = body;
-    this.extraHeaders = Utils.cloneArray(extraHeaders); // Fill the Common SIP Request Headers.
+    this.extraHeaders = Utils.cloneArray(extraHeaders);
+
+    if (this.ua.configuration.extra_headers) {
+      this.extraHeaders = this.extraHeaders.concat(this.ua.configuration.extra_headers);
+    } // Fill the Common SIP Request Headers.
     // Route.
 
+
     if (params.route_set) {
@@ -664,2 +669,7 @@ this.setHeader('route', params.route_set);
       extraHeaders = Utils.cloneArray(extraHeaders);
+
+      if (this.ua.configuration.extra_headers) {
+        extraHeaders = extraHeaders.concat(this.ua.configuration.extra_headers);
+      }
+
       var response = "SIP/2.0 ".concat(code, " ").concat(reason, "\r\n");
@@ -824,2 +834,19 @@
       response += "CSeq: ".concat(this.cseq, " ").concat(this.method, "\r\n");
+
+      if (this.ua.configuration.extra_headers) {
+        var _iterator12 = _createForOfIteratorHelper(this.ua.configuration.extra_headers),
+            _step12;
+
+        try {
+          for (_iterator12.s(); !(_step12 = _iterator12.n()).done;) {
+            var header = _step12.value;
+            response += "".concat(header.trim(), "\r\n");
+          }
+        } catch (err) {
+          _iterator12.e(err);
+        } finally {
+          _iterator12.f();
+        }
+      }
+
       response += "Content-Length: ".concat(0, "\r\n\r\n");
@@ -826,0 +853,0 @@ this.transport.send(response);

lib/Config.js

@@ -44,2 +44,5 @@ const Utils = require('./Utils');
 
+  // Global extra headers, to be added to every request and response
+  extra_headers : null,
+
   /*
@@ -312,2 +315,24 @@ * Host address.
       }
+    },
+
+    extra_headers(extra_headers)
+    {
+      const _extraHeaders = [];
+
+      if (Array.isArray(extra_headers) && extra_headers.length)
+      {
+        for (const header of extra_headers)
+        {
+          if (typeof header === 'string')
+          {
+            _extraHeaders.push(header);
+          }
+        }
+      }
+      else
+      {
+        return;
+      }
+
+      return _extraHeaders;
     }
@@ -314,0 +339,0 @@ }

lib/Constants.d.ts

@@ -5,3 +5,3 @@ export const USER_AGENT: string
 
-export enum causes {
+export declare enum causes {
   CONNECTION_ERROR = 'Connection Error',
@@ -55,3 +55,3 @@ REQUEST_TIMEOUT = 'Request Timeout',
 
-export enum DTMF_TRANSPORT {
+export declare enum DTMF_TRANSPORT {
   INFO = 'INFO',
@@ -58,0 +58,0 @@ RFC2833 = 'RFC2833',

lib/Message.d.ts

@@ -1,2 +0,1 @@
-/// <reference types="node" />
 import {EventEmitter} from 'events'
@@ -3,0 +2,0 @@

lib/RTCSession.d.ts

@@ -1,3 +0,2 @@
-/// <reference types="node" />
-import {EventEmitter} from 'events'
+import {EventEmitter, Listener} from 'events'
 
@@ -17,3 +16,3 @@ import {IncomingRequest, IncomingResponse, OutgoingRequest} from './SIPMessage'
 
-export enum SessionDirection {
+export declare enum SessionDirection {
   INCOMING = 'incoming',
@@ -23,3 +22,3 @@ OUTGOING = 'outgoing',
 
-export enum Originator {
+export declare enum Originator {
   LOCAL = 'local',
@@ -183,3 +182,2 @@ REMOTE = 'remote',
 // listener
-export type AnyListener = (...args: any[]) => void;
 export type PeerConnectionListener = (event: PeerConnectionEvent) => void;
@@ -230,7 +228,7 @@ export type ConnectingListener = (event: ConnectingEvent) => void;
   'icecandidate': IceCandidateListener;
-  'getusermediafailed': AnyListener;
-  'peerconnection:createofferfailed': AnyListener;
-  'peerconnection:createanswerfailed': AnyListener;
-  'peerconnection:setlocaldescriptionfailed': AnyListener;
-  'peerconnection:setremotedescriptionfailed': AnyListener;
+  'getusermediafailed': Listener;
+  'peerconnection:createofferfailed': Listener;
+  'peerconnection:createanswerfailed': Listener;
+  'peerconnection:setlocaldescriptionfailed': Listener;
+  'peerconnection:setremotedescriptionfailed': Listener;
 }
@@ -237,0 +235,0 @@

lib/SIPMessage.js

@@ -39,2 +39,7 @@ const sdp_transform = require('sdp-transform');
 
+    if (this.ua.configuration.extra_headers)
+    {
+      this.extraHeaders = this.extraHeaders.concat(this.ua.configuration.extra_headers);
+    }
+
     // Fill the Common SIP Request Headers.
@@ -601,2 +606,7 @@
 
+    if (this.ua.configuration.extra_headers)
+    {
+      extraHeaders = extraHeaders.concat(this.ua.configuration.extra_headers);
+    }
+
     let response = `SIP/2.0 ${code} ${reason}\r\n`;
@@ -744,2 +754,11 @@
     response += `CSeq: ${this.cseq} ${this.method}\r\n`;
+
+    if (this.ua.configuration.extra_headers)
+    {
+      for (const header of this.ua.configuration.extra_headers)
+      {
+        response += `${header.trim()}\r\n`;
+      }
+    }
+
     response += `Content-Length: ${0}\r\n\r\n`;
@@ -746,0 +765,0 @@

lib/UA.d.ts

@@ -1,6 +0,5 @@
-/// <reference types="node" />
-import {EventEmitter} from 'events'
+import {EventEmitter, Listener} from 'events'
 
 import {DisconnectEvent, Socket, WeightedSocket} from './WebSocketInterface'
-import {AnswerOptions, AnyListener, Originator, RTCSession, RTCSessionEventMap, TerminateOptions} from './RTCSession'
+import {AnswerOptions, Originator, RTCSession, RTCSessionEventMap, TerminateOptions} from './RTCSession'
 import {IncomingRequest, IncomingResponse, OutgoingRequest} from './SIPMessage'
@@ -47,2 +46,3 @@ import {Message, SendMessageOptions} from './Message'
   user_agent?: string;
+  extra_headers?: string[];
 }
@@ -128,3 +128,3 @@
   registrationFailed: RegistrationFailedListener;
-  registrationExpiring: AnyListener;
+  registrationExpiring: Listener;
   newRTCSession: RTCSessionListener;
@@ -131,0 +131,0 @@ newMessage: MessageListener;

package.json

@@ -5,3 +5,3 @@ {
   "description": "the Javascript SIP library",
-  "version": "3.9.4",
+  "version": "3.10.0",
   "homepage": "https://jssip.net",
@@ -31,4 +31,4 @@ "contributors": [
   "dependencies": {
+    "@types/events": "^3.0.0",
     "@types/debug": "^4.1.7",
-    "@types/node": "^18.11.7",
     "debug": "^4.3.1",
@@ -35,0 +35,0 @@ "events": "^3.3.0",

No alert changes

Improved metrics

Total package byte prevSize

1613321

increased by0.14%

Lines of code

46305

increased by0.17%

Dependency changes

• + Added@types/events@^3.0.0

• + Added@types/events@3.0.3(transitive)

• - Removed@types/node@^18.11.7

• - Removed@types/node@18.19.33(transitive)

• - Removedundici-types@5.26.5(transitive)