support opus transcode (#150)

* support opus transcode

* dont resend reinvite when release media for srtp

* wip

* fix review comment

lib/call-session.js

@@ -144,11 +144,6 @@ class CallSession extends Emitter {
     return this._mediaReleased;
   }
 
-  get calleeIsUsingSrtp() {
-    const tp = this.rtpEngineOpts?.uac?.mediaOpts['transport-protocol'];
-    return tp && -1 !== tp.indexOf('SAVP');
-  }
-
   subscribeForDTMF(dlg) {
     if (!this._subscribedForDTMF) {
       this._subscribedForDTMF = true;
@@ -839,7 +834,9 @@ Duration=${payload.duration} `
 
       /* if this is a re-invite from the FS to change media anchoring, avoid sending the reinvite out */
       let sdp;
-      if (isReleasingMedia && !this.calleeIsUsingSrtp) {
+      //HL 2024-11-13: previously forwarded re-invites to webrtc clients but further testing has shown to be unnecessary
+      //if (isReleasingMedia && !this.calleeIsUsingSrtp) {
+      if (isReleasingMedia) {
         this.logger.info(`got a reinvite from FS to ${reason}`);
         sdp = dlg.other.remote.sdp;
         if (!answerMedia.flags.includes('port latching')) answerMedia.flags.push('port latching');

lib/utils.js

@@ -40,7 +40,11 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams
     },
     uac: {
       tag: null,
-      mediaOpts: dstOpts
+      mediaOpts: {
+        ...dstOpts,
+        ...(process.env.JAMBONES_CODEC_OFFER_WITH_ORDER &&
+          { codec: { offer: process.env.JAMBONES_CODEC_OFFER_WITH_ORDER.split(','), strip: 'all' } }),
+      }
     }
   };
 }