wip (#155)

app.js

@@ -68,6 +68,8 @@ const {
   decrKey,
   retrieveSet,
   isMemberOfSet,
+  addKey,
+  retrieveKey
 } = require('@jambonz/realtimedb-helpers')({}, logger);
 
 const activeCallIds = new Map();
@@ -101,6 +103,8 @@ srf.locals = {...srf.locals,
   },
   realtimeDbHelpers: {
     client: redisClient,
+    addKey,
+    retrieveKey,
     createHash,
     retrieveHash,
     incrKey,

lib/call-session.js

@@ -1,7 +1,15 @@
 const Emitter = require('events');
 const sdpTransform = require('sdp-transform');
 const SrsClient = require('@jambonz/siprec-client-utils');
-const {makeRtpEngineOpts, nudgeCallCounts, isPrivateVoipNetwork, isBlackListedSipGateway} = require('./utils');
+const {
+  makeRtpEngineOpts,
+  nudgeCallCounts,
+  isPrivateVoipNetwork,
+  isBlackListedSipGateway,
+  makeFullMediaReleaseKey,
+  makePartnerFullMediaReleaseKey
+} = require('./utils');
+const { MediaPath } = require('./constants.json');
 const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
 const {SipError, stringifyUri, parseUri} = require('drachtio-srf');
 const debug = require('debug')('jambonz:sbc-outbound');
@@ -111,6 +119,8 @@ class CallSession extends Emitter {
     this.writeCdrs = this.srf.locals.writeCdrs;
 
     this.decrKey = req.srf.locals.realtimeDbHelpers.decrKey;
+    this.addKey = req.srf.locals.realtimeDbHelpers.addKey;
+    this.retrieveKey = req.srf.locals.realtimeDbHelpers.retrieveKey;
 
     const {
       lookupOutboundCarrierForAccount,
@@ -123,7 +133,8 @@ class CallSession extends Emitter {
     this.lookupSipGatewaysByCarrier = lookupSipGatewaysByCarrier;
     this.lookupCarrierByAccountLcr = lookupCarrierByAccountLcr;
 
-    this._mediaReleased = false;
+    this._mediaPath = MediaPath.FullMedia;
+
     this.recordingNoAnswerTimeout = (process.env.JAMBONES_RECORDING_NO_ANSWER_TIMEOUT || 2) * 1000;
   }
 
@@ -144,7 +155,7 @@ class CallSession extends Emitter {
   }
 
   get isMediaReleased() {
-    return this._mediaReleased;
+    return this._mediaPath !== MediaPath.FullMedia;
   }
 
   subscribeForDTMF(dlg) {
@@ -651,6 +662,15 @@ class CallSession extends Emitter {
         answered_at: callStart
       };
     }
+
+    /* save far end SDP for later use if we do a full media release */
+    if (process.env.JAMBONES_ENABLE_FULL_MEDIA_RELEASE) {
+      const key = makeFullMediaReleaseKey(this.req.get('X-CID'));
+      const sdp = uac.remote.sdp;
+      this.logger.info({key, sdp}, 'saving far end sdp for full media release feature');
+      this.addKey(key, sdp, 3600).catch((err) => this.logger.error(err, 'Error saving far end sdp'));
+    }
+
     this.uas = uas;
     this.uac = uac;
     [uas, uac].forEach((dlg) => {
@@ -807,11 +827,32 @@ Duration=${payload.duration} `
     try {
       const reason = req.get('X-Reason');
       const isReleasingMedia = reason && dlg.type === 'uas' && ['release-media', 'anchor-media'].includes(reason);
+      const isFullMediaRelease = reason === 'release-media-entirely' && process.env.JAMBONES_ENABLE_FULL_MEDIA_RELEASE;
       const fromTag = dlg.type === 'uas' ? this.rtpEngineOpts.uas.tag : this.rtpEngineOpts.uac.tag;
       const toTag = dlg.type === 'uas' ? this.rtpEngineOpts.uac.tag : this.rtpEngineOpts.uas.tag;
       const offerMedia = dlg.type === 'uas' ? this.rtpEngineOpts.uac.mediaOpts : this.rtpEngineOpts.uas.mediaOpts;
       const answerMedia = dlg.type === 'uas' ? this.rtpEngineOpts.uas.mediaOpts : this.rtpEngineOpts.uac.mediaOpts;
       const direction =  dlg.type === 'uas' ? ['private', 'public'] : ['public', 'private'];
+
+      if (isFullMediaRelease) {
+        const a_sdp = await this.retrieveKey(makePartnerFullMediaReleaseKey(this.req.get('X-CID')));
+        this.logger.info({a_sdp}, 'reinvite ourselves out of the media path with this reinvite offer');
+        const answerSdp = await dlg.other.modify(a_sdp);
+        this.logger.info({answerSdp}, 'far end response to full media release');
+        res.send(200, {
+          body: dlg.local.sdp,
+          headers: {
+            'Contact': this.contactHeader
+          }
+        });
+        /* no media going through us now we can destroy the rtpengine resource */
+        this.rtpEngineResource.destroy().catch((err) => {
+          this.logger.info({err}, 'Error destroying rtpengine resource after full media release');
+        });
+        this._mediaPath = MediaPath.NoMedia;
+        return;
+      }
+
       if (isReleasingMedia) {
         if (!offerMedia.flags.includes('port latching')) offerMedia.flags.push('port latching');
         if (!offerMedia.flags.includes('asymmetric')) offerMedia.flags.push('asymmetric');
@@ -839,17 +880,18 @@ Duration=${payload.duration} `
       let sdp;
       //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) {
+
+      //DH 2024-11- 18: if we are going from no-media to either partial or full media, we need reinvite the far end
+      if (isReleasingMedia && this._mediaPath !== MediaPath.NoMedia) {
         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');
         if (!answerMedia.flags.includes('asymmetric')) answerMedia.flags.push('asymmetric');
         answerMedia.flags = answerMedia.flags.filter((f) => f !== 'media handover');
-        this._mediaReleased = 'release-media' === reason;
+        this._mediaPath = 'release-media' === reason ? MediaPath.PartialMedia : MediaPath.FullMedia;
       }
       else {
         sdp = await dlg.other.modify(response.sdp);
-        this.logger.info({sdp}, 'CallSession:_onReinvite: got sdp from 200 OK to invite we sent');
       }
       opts = {
         ...this.rtpEngineOpts.common,

lib/constants.json

@@ -0,0 +1,7 @@
+{
+  "MediaPath": {
+  "NoMedia": "no-media",
+  "PartialMedia": "partial-media",
+  "FullMedia": "full-media"
+  }
+}

lib/utils.js

@@ -233,6 +233,13 @@ async function isBlackListedSipGateway(client, logger, sip_gateway_sid) {
   }
 }
 
+const makeFullMediaReleaseKey = (callId) => {
+  return `b_sdp:${callId}`;
+};
+const makePartnerFullMediaReleaseKey = (callId) => {
+  return `a_sdp:${callId}`;
+};
+
 module.exports = {
   makeRtpEngineOpts,
   selectHostPort,
@@ -244,5 +251,7 @@ module.exports = {
   createHealthCheckApp,
   nudgeCallCounts,
   isPrivateVoipNetwork,
-  isBlackListedSipGateway
+  isBlackListedSipGateway,
+  makeFullMediaReleaseKey,
+  makePartnerFullMediaReleaseKey
 };