Fresh/466 (#160 )

* Hold and unhold does not resume transcript for outbound call to webrtc * WIP * jslint * jslint * handle null * Added direction * re-invite with opus codec need to change pcmu on reinvite on unhold opus codec was been sent from rtpegine to freeswitch due to which transcripts were not getting generated , hence it needs to be changed to pcmu * jslint jslint
support referby display name (#161 )
2026-10-04 18:44:38 +00:00 · 2024-12-23 07:20:04 -05:00 · 2024-12-11 10:34:27 -05:00 · 2024-12-05 21:16:12 -05:00 · 2024-11-22 10:46:24 -05:00 · 2024-11-19 09:37:53 -05:00
8 changed files with 120 additions and 36 deletions
@@ -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,
@@ -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');
@@ -89,7 +97,10 @@ const updateRtpEngineFlags = (sdp, opts) => {
  try {
    const parsed = sdpTransform.parse(sdp);
    const codec = parsed.media[0].rtp[0].codec;
-    if (['PCMU', 'PCMA'].includes(codec)) opts.flags.push(`codec-accept-${codec}`);
+    if (['PCMU', 'PCMA'].includes(codec)) {
+      opts.flags.push(`codec-accept-${codec}`);
+      opts.flags.push('codec-accept-telephone-event');
+    }
  } catch (err) {}
  return opts;
 };
@@ -108,6 +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,
@@ -120,7 +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;
  }

@@ -141,12 +155,7 @@ class CallSession extends Emitter {
  }

  get isMediaReleased() {
-    return this._mediaReleased;
-  }
-
-  get calleeIsUsingSrtp() {
-    const tp = this.rtpEngineOpts?.uac?.mediaOpts['transport-protocol'];
-    return tp && -1 !== tp.indexOf('SAVP');
+    return this._mediaPath !== MediaPath.FullMedia;
  }

  subscribeForDTMF(dlg) {
@@ -653,6 +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) => {
@@ -668,7 +686,6 @@ class CallSession extends Emitter {
            if (!IMMUTABLE_HEADERS.includes(h)) headers[h] = bye.headers[h];
          });
          await other.destroy({headers});
-          await other.destroy();
        } catch (err) {}

        const trackingOn = process.env.JAMBONES_TRACK_ACCOUNT_CALLS ||
@@ -677,12 +694,16 @@ class CallSession extends Emitter {

        if (process.env.JAMBONES_HOSTING || trackingOn) {
          const {writeCallCount, writeCallCountSP, writeCallCountApp} = this.req.srf.locals;
-          await nudgeCallCounts(this.logger, {
-            service_provider_sid: this.service_provider_sid,
-            account_sid: this.account_sid,
-            application_sid: this.application_sid
-          }, this.decrKey, {writeCallCountSP, writeCallCount, writeCallCountApp})
-            .catch((err) => this.logger.error(err, 'Error decrementing call counts'));
+          if (!this.req.locals.callCountNudged) {
+            this.req.locals.callCountNudged = true;
+            this.logger.debug('decrementing call count at end of call');
+            await nudgeCallCounts(this.logger, {
+              service_provider_sid: this.service_provider_sid,
+              account_sid: this.account_sid,
+              application_sid: this.application_sid
+            }, this.decrKey, {writeCallCountSP, writeCallCount, writeCallCountApp})
+              .catch((err) => this.logger.error(err, 'Error decrementing call counts'));
+          }
        }

        /* write cdr for connected call */
@@ -759,12 +780,19 @@ class CallSession extends Emitter {
  async _onRefer(dlg, req, res) {
    /* REFER coming in from a sip device, forward to feature server */
    try {
+      const customHeaders = Object.keys(req.headers)
+        .filter((h) => h.toLowerCase().startsWith('x-'))
+        .reduce((acc, h) => {
+          acc[h] = req.get(h);
+          return acc;
+        }, {});
      const response = await dlg.other.request({
        method: 'REFER',
        headers: {
          'Refer-To': req.get('Refer-To'),
          'Referred-By': req.get('Referred-By'),
-          'User-Agent': req.get('User-Agent')
+          'User-Agent': req.get('User-Agent'),
+          ...customHeaders
        }
      });
      res.send(response.status, response.reason);
@@ -805,16 +833,44 @@ 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;
+      let offerMedia = dlg.type === 'uas' ? this.rtpEngineOpts.uac.mediaOpts : this.rtpEngineOpts.uas.mediaOpts;
+      let 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');
        offerMedia.flags = offerMedia.flags.filter((f) => f !== 'media handover');
      }
+      const acceptCodecs = process.env.JAMBONES_ACCEPT_AND_TRANSCODE ?
+        process.env.JAMBONES_ACCEPT_AND_TRANSCODE :
+        process.env.JAMBONES_ACCEPT_G729 ? 'g729' : '';
+      offerMedia = { ...offerMedia, ...(acceptCodecs && dlg.type === 'uac' && 
+        { codec: { mask: acceptCodecs, transcode: 'pcmu,pcma' } })};
+      answerMedia = { ...answerMedia, ...(process.env.JAMBONES_CODEC_OFFER_WITH_ORDER && dlg.type === 'uac' && 
+        { codec: {offer: process.env.JAMBONES_CODEC_OFFER_WITH_ORDER.split(','), strip: 'all' }})};
      let opts = {
        ...this.rtpEngineOpts.common,
        ...offerMedia,
@@ -835,17 +891,20 @@ 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) {
+
+      //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 ${reason}`);
        sdp = dlg.other.remote.sdp;
        if (!answerMedia.flags.includes('port latching')) answerMedia.flags.push('port latching');
        if (!answerMedia.flags.includes('asymmetric')) answerMedia.flags.push('asymmetric');
        answerMedia.flags = answerMedia.flags.filter((f) => f !== 'media handover');
-        this._mediaReleased = 'release-media' === reason;
+        this._mediaPath = 'release-media' === reason ? MediaPath.PartialMedia : MediaPath.FullMedia;
      }
      else {
        sdp = await dlg.other.modify(response.sdp);
-        this.logger.info({sdp}, 'CallSession:_onReinvite: got sdp from 200 OK to invite we sent');
      }
      opts = {
        ...this.rtpEngineOpts.common,
@@ -1094,7 +1153,7 @@ Duration=${payload.duration} `
      // eslint-disable-next-line no-unused-vars
      const { via, from, to, 'call-id': callid, cseq, 'max-forwards': maxforwards,
      // eslint-disable-next-line no-unused-vars
-        'content-length': _contentlength, 'refer-to': _referto, 'referred-by': _referredby,
+        'content-length': _contentlength, 'refer-to': _referto, 'referred-by': referredby,
        // eslint-disable-next-line no-unused-vars
        'X-Refer-To-Leave-Untouched': _leave,
        ...customHeaders
@@ -1106,9 +1165,7 @@ Duration=${payload.duration} `
      const arr = /context-(.*)/.exec(uri.user);
      if (!arr) {
        /* call transfer requested */
-        const referredBy = req.getParsedHeader('Referred-By');
-        if (!referredBy) return res.send(400);
-        const u = parseUri(referredBy.uri);
+        if (!req.has('Referred-By')) return res.send(400);

        /* delete contact if it was there from feature server */
        delete customHeaders['contact'];
@@ -1117,7 +1174,7 @@ Duration=${payload.duration} `
          method: 'REFER',
          headers: {
            'Refer-To': `<${stringifyUri(uri)}>`,
-            'Referred-By': `<${stringifyUri(u)}>`,
+            'Referred-By': referredby,
            ...customHeaders
          }
        });
@@ -0,0 +1,7 @@
+{
+  "MediaPath": {
+  "NoMedia": "no-media",
+  "PartialMedia": "partial-media",
+  "FullMedia": "full-media"
+  }
+}
@@ -106,7 +106,9 @@ module.exports = (srf, logger, redisClient) => {
    try {
      /* decrement count if INVITE is later rejected */
      res.once('end', async({status}) => {
-        if (status > 200) {
+        if (status > 200 && !req.locals.callCountNudged) {
+          req.locals.callCountNudged = true;
+          logger.debug('decrementing call count due to call rejection');
          nudgeCallCounts(logger, {
            service_provider_sid,
            account_sid,
@@ -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' } }),
+      }
    }
  };
 }
@@ -229,6 +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,
@@ -240,5 +251,7 @@ module.exports = {
  createHealthCheckApp,
  nudgeCallCounts,
  isPrivateVoipNetwork,
-  isBlackListedSipGateway
+  isBlackListedSipGateway,
+  makeFullMediaReleaseKey,
+  makePartnerFullMediaReleaseKey
 };
@@ -14,7 +14,7 @@
        "@jambonz/mw-registrar": "0.2.7",
        "@jambonz/realtimedb-helpers": "^0.8.9",
        "@jambonz/rtpengine-utils": "^0.4.4",
-        "@jambonz/siprec-client-utils": "^0.2.7",
+        "@jambonz/siprec-client-utils": "^0.2.9",
        "@jambonz/stats-collector": "^0.1.10",
        "@jambonz/time-series": "^0.2.10",
        "cidr-matcher": "^2.1.1",
@@ -628,9 +628,10 @@
      }
    },
    "node_modules/@jambonz/siprec-client-utils": {
-      "version": "0.2.7",
-      "resolved": "https://registry.npmjs.org/@jambonz/siprec-client-utils/-/siprec-client-utils-0.2.7.tgz",
-      "integrity": "sha512-VztOToBfXnOphg/y6kO+lBYqoDH7X5Ci0daXT5zQ8v5NnaZw+UwxGeFPX3knIR3r5pmzUld0f2giFOuD4ZOF8w==",
+      "version": "0.2.9",
+      "resolved": "https://registry.npmjs.org/@jambonz/siprec-client-utils/-/siprec-client-utils-0.2.9.tgz",
+      "integrity": "sha512-C/1UOrAvQVG0iwiSMS0MC9VHRjhg3/sOjNna+eEjHjws1SHl8NrKBoDL8uOwLqOM2ccpbqCPH9GJAxo5OnOhww==",
+      "license": "MIT",
      "dependencies": {
        "sdp-transform": "^2.14.1",
        "uuid": "^8.3.2"
@@ -32,7 +32,7 @@
    "@jambonz/http-health-check": "^0.0.1",
    "@jambonz/mw-registrar": "0.2.7",
    "@jambonz/rtpengine-utils": "^0.4.4",
-    "@jambonz/siprec-client-utils": "^0.2.7",
+    "@jambonz/siprec-client-utils": "^0.2.9",
    "@jambonz/stats-collector": "^0.1.10",
    "@jambonz/time-series": "^0.2.10",
    "cidr-matcher": "^2.1.1",
@@ -24,7 +24,7 @@ services:
        ipv4_address: 172.39.0.2

  sbc:
-    image: drachtio/drachtio-server:latest
+    image: drachtio/drachtio-server:0.8.26
    command: drachtio --contact "sip:*;transport=udp" --loglevel debug --sofia-loglevel 9
    ports:
      - "9060:9022/tcp"
Author	SHA1	Message	Date
Vinod Dharashive	d8ae824559	Fresh/466 (#160 ) * Hold and unhold does not resume transcript for outbound call to webrtc * WIP * jslint * jslint * handle null * Added direction * re-invite with opus codec need to change pcmu on reinvite on unhold opus codec was been sent from rtpegine to freeswitch due to which transcripts were not getting generated , hence it needs to be changed to pcmu * jslint jslint	2024-12-23 07:20:04 -05:00
Hoan Luu Huu	334db6f84f	support referby display name (#161 )	2024-12-11 10:34:27 -05:00
Hoan Luu Huu	c762accce8	forward all refer to feature server (#156 ) * forward all refer to feature server * forwawrd extra custom header	2024-12-05 21:16:12 -05:00
Dave Horton	c6c63e26da	fix duplicate attempt to destroy dialog	2024-11-22 10:46:24 -05:00
Dave Horton	916d577b75	wip (#155 )	2024-11-19 09:37:53 -05:00
Dave Horton	b976a62a60	when forcing PCMU or PCMA we must also include telephone-event (#153 )	2024-11-14 08:37:40 -05:00
Hoan Luu Huu	5e2369e5e8	support opus transcode (#150 ) * support opus transcode * dont resend reinvite when release media for srtp * wip * fix review comment	2024-11-13 07:41:43 -05:00
Dave Horton	41528db630	fix call count race condition #151 (#152 ) * fix call count race condition #151 * logging fix	2024-11-12 14:57:04 -05:00