@babblevoice/babble-drachtio-callmanager 3.5.6 → 3.5.7

package/lib/call.js

@@ -113,18 +113,6 @@ class SipError extends Error {
   }
 }
 
-/* Reverse codes - include inbound error codes.
-If not in this list we return REQUEST_TERMINATED during creation */
-/*
-const inboundsiperros = {
-  486: hangupcodes.USER_BUSY,
-  408: hangupcodes.REQUEST_TIMEOUT,
-  404: hangupcodes.UNALLOCATED_NUMBER,
-  603: hangupcodes.DECLINED
-}
-*/
-
-
 const inboundsiperros = {}
 const hangupcodeskeys = Object.keys( hangupcodes )
 for (let i = 0; i < hangupcodeskeys.length; i++) {
@@ -219,6 +207,13 @@ class call {
       "closed": {
         "audio": []
       },
+      "secure": {
+        "audio": {
+          "offer": true,
+          "dtlscontroller": false,
+          "reinvite": false
+        }
+      },
       "count": 0
     }
 
@@ -1201,6 +1196,7 @@ class call {
 
     /* we have this._res */
     this.sdp.remote = sdpgen.create( this._res.msg.body )
+    this.#parsesrtpsetup()
     await this.answer( { "early": true } )
 
     if( !this.parent || this.parent.state.established ) return
@@ -1269,6 +1265,8 @@ class call {
       return this.hangup( hangupcodes.INCOMPATIBLE_DESTINATION )
     }
 
+    this.#parsesrtpsetup()
+
     this.sdp.remote.setdynamepayloadtypes( this.sdp.local )
 
     const channeldef = await this.#createchannelremotedef()
@@ -1294,6 +1292,8 @@ class call {
     else
       this.sdp.remote = sdpgen.create( this._req.msg.body )
 
+    this.#parsesrtpsetup()
+
     const selectedcodec = this.sdp.remote.intersection( this.options.preferedcodecs, true )
     if( !selectedcodec ) {
       return this.hangup( hangupcodes.INCOMPATIBLE_DESTINATION )
@@ -1321,7 +1321,7 @@ class call {
     if( this._iswebrtc ) {
       const ch = this.channels.audio
       this.sdp.local.addssrc( ch.local.ssrc )
-        .secure( ch.local.dtls.fingerprint, channeldef.remote.dtls.mode )
+        .secure( ch.local.dtls.fingerprint, this.#getsrtpsetup( this.channels.secure.audio ) )
         .addicecandidates( ch.local.address, ch.local.port, ch.local.icepwd )
         .rtcpmux()
     }
@@ -1651,7 +1651,7 @@ class call {
 
       if( this._iswebrtc ) {
         this.sdp.local.addssrc( this.channels.audio.local.ssrc )
-          .secure( this.channels.audio.local.dtls.fingerprint, channeldef.remote.dtls.mode )
+          .secure( this.channels.audio.local.dtls.fingerprint, this.#getsrtpsetup( this.channels.secure.audio ) )
           .addicecandidates( this.channels.audio.local.address, this.channels.audio.local.port, this.channels.audio.local.icepwd )
           .rtcpmux()
       }
@@ -2197,6 +2197,8 @@ class call {
 
   #getsdpformodify( req ) {
     const sdp = sdpgen.create( req.msg.body )
+
+    this.#parsesrtpsetup( sdp )
     const media = sdp.getmedia()
 
     let ip
@@ -2267,8 +2269,7 @@ class call {
     by AWOL... */
     this._timers.seinterval = setInterval( async () => {
 
-      try{
-
+      try {
         if( this.destroyed ) {
           /* this should be done - but we are still running */
           clearInterval( this._timers.seinterval )
@@ -2279,8 +2280,9 @@ class call {
 
         if( this._iswebrtc && this.channels && this.channels.audio ) {
           /* we are sending invite so we MUST offer be actpass */
+          this.channels.secure.audio.offer = true
           this.sdp.local
-            .secure( this.channels.audio.local.dtls.fingerprint ,"actpass" )
+            .secure( this.channels.audio.local.dtls.fingerprint, this.#getsrtpsetup( this.channels.secure.audio ) )
         }
 
         const opts = {
@@ -2334,17 +2336,14 @@ class call {
       //  The application must respond, using the res parameter provided.
       if( "INVITE" !== req.msg.method ) return
 
+      this.channels.secure.audio.reinvite = true
+      const sdp = this.#getsdpformodify( req )
+
       if( this._iswebrtc && this.channels && this.channels.audio ) {
-        /* I cannot figure out these rules - after reading RFC 3264 and 4145
-           I have fixed it like this as this is required by Chrome - but I am unsure why it is demanding this */
-        let mode = "passive"
-        if( "uas" == this.type ) mode = "active"
         this.sdp.local
-          .secure( this.channels.audio.local.dtls.fingerprint, mode )
+          .secure( this.channels.audio.local.dtls.fingerprint, this.#getsrtpsetup( this.channels.secure.audio ) )
       }
 
-      const sdp = this.#getsdpformodify( req )
-
       /* this was tested against jssip - which I don't think is correct. It was sending us
       sendonly when placing the call on hold. It didn't change the connection IP (although it did set the rtcp connection ip to 0.0.0.0!). */
       let d = sdp.media.direction
@@ -2634,6 +2633,59 @@ class call {
     return target
   }
 
+  /**
+   * I cannot figure out these rules - after reading RFC 3264 and 4145
+   * I have fixed it like this as this is required by Chrome - but I am unsure why it is demanding this
+   * This is for our local SDP - i.e. from our perspective.
+   * @param { object } secure - the secure object i.e. this.channels.secure.audio
+   * @returns { "passive" | "active" | "actpass" }
+   */
+  #getsrtpsetup( secure ) {
+    if( secure.offer ) {
+      /* ref: https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-jsep-14#page-34 offer MUST be actpass */
+      secure.dtlscontroller = false
+      secure.offer = false
+      return "actpass"
+    }
+
+    return secure.dtlscontroller ? "active" : "passive"
+  }
+
+  /**
+   * Parses our remote SDP to decide if we are the controller or not.
+   * TODO - parse SDP for other streams.
+   * @param { object } [ sdp ] - the sdp object - or leave if use our remote
+   * @returns 
+   */
+  #parsesrtpsetup( sdp ) {
+    if( !sdp ) sdp = this.sdp.remote
+    if( !sdp ) return
+
+    const audio = sdp.getmedia( "audio" )
+    if( !audio ) return
+
+    if( "passive" == audio.setup ||
+        ( "actpass" == audio.setup && !this.channels.secure.audio.reinvite ) ) {
+      /* we might also have to re-negotiate our rtp server - but this should keep it in the right direction */
+      this.channels.secure.audio.dtlscontroller = true
+      this.channels.secure.audio.reinvite = false
+    } else if( "active" == audio.setup ) {
+      this.channels.secure.audio.dtlscontroller = false
+    }
+
+    this.channels.secure.audio.offer = false
+  }
+
+  /**
+   * Get the mode string for our media server perspective.
+   * @param { object } media object returned by sdp.getmedia()
+   * @returns { "active" | "passive" }
+   */
+  #getsrtpsetupforchannel( media ) {
+    if( "active" == media.setup ) return "passive"
+    else return "active"
+  }
+
   /**
    * Returns and object we can pass into an openchannel function, 
    * { remote: {} } - the remote can be passed into set remote on
@@ -2663,14 +2715,14 @@ class call {
 
     if( iswebrtc ) {
 
-      const hash = this.sdp.remote.sdp.media[ 0 ].fingerprint.hash
-      let mode = "active"
-      if( "active" == this.sdp.remote.sdp.media[ 0 ].setup ) mode = "passive" /* act|pass|actpass */
-  
+      const audio = this.sdp.remote.getmedia( "audio" )
+      if( !audio ) return
+
+      const hash = audio.fingerprint.hash
       if( hash ) {
         chandef.remote.dtls = {
           "fingerprint": { hash },
-          mode
+          "mode": this.#getsrtpsetupforchannel( audio )
         }
       }
     }
@@ -3104,10 +3156,10 @@ class call {
 
       /* DTLS is only supported ( outbound ) on websocket connections */
       if( this._iswebrtc ) {
+        this.channels.secure.audio.offer = true
         this.sdp.local
           .addssrc( this.channels.audio.local.ssrc )
-          /* ref: https://datatracker.ietf.org/doc/html/draft-ietf-rtcweb-jsep-14#page-34 offer MUST be actpass */
-          .secure( this.channels.audio.local.dtls.fingerprint ,"actpass" )
+          .secure( this.channels.audio.local.dtls.fingerprint, this.#getsrtpsetup( this.channels.secure.audio ) )
           .addicecandidates( this.channels.audio.local.address, this.channels.audio.local.port, this.channels.audio.local.icepwd )
           .rtcpmux()
       }
@@ -3550,6 +3602,7 @@ class call {
     if( c._req.msg && c._req.msg.body ) {
       c.sdp.remote = sdpgen.create( c._req.msg.body )
       c.sdp.remote.intersection( callmanager.options.preferedcodecs, true )
+      c.#parsesrtpsetup()
     }
 
     /* set a timer using the default */

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "@babblevoice/babble-drachtio-callmanager",
-  "version": "3.5.6",
+  "version": "3.5.7",
   "description": "Call processing to create a PBX",
   "main": "index.js",
   "scripts": {