0.9.11

* fix: keep siprec recording alive across a feature server transfer

A cross-feature-server move (enqueue/dequeue or conference) re-negotiates the
feature-server leg in _onFeatureServerTransfer, but nothing rebuilt the rtpengine
subscription the SIPREC recording forks from, so the recorder went silent from the
moment the call moved. The fresh destroy handler installed on the new leg also
dropped the _stopRecording() call the original handlers have, so the SIPREC dialog
was never BYEd and the recorder had to wait out its media timeout.

Rebuild the subscription after the transfer re-negotiates media, and stop the
recording when the transferred leg ends. The resubscribe call is guarded so this
is safe to deploy before @jambonz/siprec-client-utils is bumped.

Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>

* fix: drop the siprec re-subscribe, keep the missing teardown

The cross-feature-server move was tested on a real two-feature-server cluster
(inbound path): the rtpengine subscription survives the REFER on its own, so
resubscribe() was fixing a fault that does not exist. rtpengine keeps
non-offer-answer subscriptions across an answer.

What does fail, and what this branch still fixes, is the teardown: the destroy
handler installed on the transferred leg never stopped the recording, so the
recorder was left without a BYE. Verified on the inbound path; the outbound
transfer path has the identical handler and the same omission.

Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com>

---------

Co-authored-by: Claude Opus 5 (1M context) <noreply@anthropic.com>

.gitignore

@@ -34,6 +34,7 @@ build/Release
 node_modules
 
 .DS_Store
+.vscode
 
 examples/*
 CLAUDE.md

README.md

@@ -29,9 +29,19 @@ Configuration is provided via environment variables:
 |JAMBONES_RTPENGINES| commans-separated list of ip:ng-port for rtpengines (e.g. '172.31.32.10:22222')|yes|
 |JAMBONES_TIME_SERIES_HOST| influxdb host |yes|
 |JAMBONES_RECORD_ALL_CALLS| enable auto record calls |no|
+|JAMBONES_CODEC_OFFER_WITH_ORDER| comma-separated codec list to use as the outbound offer toward the carrier; the original codecs from the feature server are stripped first (e.g. `'opus,PCMU,PCMA,telephone-event'`) |no|
+|JAMBONES_CODEC_TRANSCODE| comma-separated codec list that rtpengine should add to the outbound offer and transcode on the fly when the carrier selects them — used to support codecs the feature server does not speak (e.g. `'AMR-WB/16000'`). Requires an rtpengine build that includes the corresponding codec module. |no|
+|JAMBONES_ACCEPT_AND_TRANSCODE| comma-separated codec list to accept on the outbound leg and transcode to PCMU/PCMA toward the feature server |no|
+|JAMBONES_ACCEPT_G729| if set, accept G.729 on the outbound leg and transcode to PCMU/PCMA (shorthand for `JAMBONES_ACCEPT_AND_TRANSCODE=g729`) |no|
 |K8S| service running as kubernetes service |no|
 |K8S_RTPENGINE_SERVICE_NAME| rtpengine service name(required for K8S) |no|
 
+#### Codec transcoding notes
+
+`JAMBONES_CODEC_OFFER_WITH_ORDER` and `JAMBONES_CODEC_TRANSCODE` can be combined. When both are set, rtpengine first strips the original codecs from the SDP, then adds the codecs listed in `JAMBONES_CODEC_OFFER_WITH_ORDER` to the offer, and finally adds the codecs from `JAMBONES_CODEC_TRANSCODE` with transcoding enabled. When only `JAMBONES_CODEC_TRANSCODE` is set, the original codecs from the feature server are kept in the offer and the transcode codecs are appended.
+
+Codecs that require licensed or optional rtpengine modules (AMR, AMR-WB, G.729, etc.) only work if rtpengine was compiled with support for them. Carrier-specific fmtp parameters (for example, `octet-align=1` for AMR-WB) can be appended to the codec name as needed (e.g. `AMR-WB/16000;octet-align=1`).
+
 ### running under pm2
 Typically, this application runs under [pm2](https://pm2.io) using an [ecosystem.config.js](https://pm2.keymetrics.io/docs/usage/application-declaration/) file similar to this:
 ```js

data/srtp-transcoding.json

@@ -4,7 +4,7 @@
     "ICE": "force",
     "SDES": "off",
     "flags": ["generate mid", "SDES-no", "port latching"],
-    "rtcp-mux": ["require"]  
+    "rtcp-mux": ["offer"]
   },
   "teams": {
     "transport-protocol": "RTP/SAVP",

lib/call-session.js

@@ -1,5 +1,4 @@
 const Emitter = require('events');
-const sdpTransform = require('sdp-transform');
 const SrsClient = require('@jambonz/siprec-client-utils');
 const {
   makeRtpEngineOpts,
@@ -10,7 +9,6 @@ const {
   makePartnerFullMediaReleaseKey,
   isValidDomainOrIP,
   removeVideoSdp,
-  determineAnswerCodec
 } = require('./utils');
 const { MediaPath } = require('./constants.json');
 const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
@@ -96,19 +94,6 @@ const initCdr = (req, invite) => {
   };
 };
 
-const updateRtpEngineFlags = (sdp, opts) => {
-  try {
-    const parsed = sdpTransform.parse(sdp);
-    const codec = parsed.media[0].rtp[0].codec;
-    // Only add telephone-event support, don't restrict to specific G.711 codec yet
-    // This allows far end to choose between PCMU/PCMA freely
-    if (['PCMU', 'PCMA'].includes(codec)) {
-      opts.flags.push('codec-accept-telephone-event');
-    }
-  } catch {}
-  return opts;
-};
-
 class CallSession extends Emitter {
   constructor(logger, req, res) {
     super();
@@ -125,6 +110,7 @@ class CallSession extends Emitter {
     this.decrKey = req.srf.locals.realtimeDbHelpers.decrKey;
     this.addKey = req.srf.locals.realtimeDbHelpers.addKey;
     this.retrieveKey = req.srf.locals.realtimeDbHelpers.retrieveKey;
+    this.retrieveHash = req.srf.locals.realtimeDbHelpers.retrieveHash;
 
     const {
       lookupOutboundCarrierForAccount,
@@ -251,6 +237,24 @@ class CallSession extends Emitter {
         this.logger.info(`sending call to registered user ${destUri}`);
       }
       else if (this.req.locals.target === 'forward') {
+        /* the feature-server sets X-Jambonz-SRTP (from the dial verb's srtpEncryption
+           option) to request encrypted media on a per-call basis */
+        const srtpMode = this.req.get('X-Jambonz-SRTP');
+        if (srtpMode) {
+          /* SDES (RTP/SAVP): pass teams=true to select the SDES srtp profile, matching
+             the proven carrier tls/srtp path above. This offers a=crypto to the target
+             (what SIP endpoints such as LiveKit expect), not DTLS-SRTP. */
+          this.logger.info({uri: this.req.uri, srtpMode},
+            'using SRTP (SDES) for forwarded call per X-Jambonz-SRTP');
+          this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, false, true);
+          encryptedMedia = true;
+        }
+        else if (process.env.JAMBONES_SIPS_FORWARD_SRTP && this.req.uri.startsWith('sips:')) {
+          /* legacy global opt-in for sips: forwards (uses the DTLS srtp profile) */
+          this.logger.info({uri: this.req.uri}, 'using SRTP/TLS for forwarded sips: call');
+          this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, false, false);
+          encryptedMedia = true;
+        }
         uris = [{
           private_network: await isPrivateVoipNetwork(this.req.uri),
           uri: this.req.uri
@@ -366,7 +370,8 @@ class CallSession extends Emitter {
                  * like a rare use case -- encryption is usually an all or nothing requirement.
                  */
                 this.logger.info({u}, `using SRTP for outbound call, pad crypto: ${o.pad_crypto ? 'yes' : 'no'}`);
-                this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, o.pad_crypto, true);
+                // eslint-disable-next-line max-len
+                this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, o.pad_crypto, true, o.remove_ice, o.dtls_off);
                 encryptedMedia = true;
               }
             });
@@ -398,13 +403,13 @@ class CallSession extends Emitter {
       const toPublic = uris.some((u) => u.private_network === false);
       let isOfferUpdatedToPrivate = toPrivate && !toPublic;
 
-      const opts = updateRtpEngineFlags(this.req.body, {
+      const opts = {
         ...this.rtpEngineOpts.common,
         ...this.rtpEngineOpts.uac.mediaOpts,
         'from-tag': this.rtpEngineOpts.uas.tag,
         direction:  ['private', toPublic ? 'public' : 'private'],
         sdp: this.req.body
-      });
+      };
       let response = await this.offer(opts);
       this.logger.debug({offer: opts, response}, 'initial offer to rtpengine');
       if ('ok' !== response.result) {
@@ -516,7 +521,7 @@ class CallSession extends Emitter {
             })
           };
           const p = proxy ? ` via ${proxy}` : '';
-          this.logger.info({uri, p}, `sending INVITE to ${uri}${p}`);
+          this.logger.info({uri, p}, `sending INVITE${p}`);
         }
 
         /* now launch an outbound call attempt */
@@ -539,6 +544,7 @@ class CallSession extends Emitter {
               '-X-Preferred-From-Host',
               '-X-Jambonz-FS-UUID',
               '-X-Voip-Carrier-Sid',
+              '-X-Jambonz-SRTP',
               '-X-SIP-Proxy'
             ],
             proxyResponseHeaders: [
@@ -557,15 +563,16 @@ class CallSession extends Emitter {
             localSdpA: async(sdp, res) => {
               this.rtpEngineOpts.uac.tag = res.getParsedHeader('To').params.tag;
 
-              // Determine which codec to use based on far end negotiation
-              const {codec} = determineAnswerCodec(sdp, this.req.body, this.logger);
-
               const opts = {
                 ...this.rtpEngineOpts.common,
                 ...this.rtpEngineOpts.uas.mediaOpts,
                 'from-tag': this.rtpEngineOpts.uas.tag,
                 'to-tag': this.rtpEngineOpts.uac.tag,
-                flags: ['single codec', 'inject DTMF', `codec-accept-${codec}`, 'codec-accept-telephone-event'],
+                flags: [
+                  'single codec',
+                  'inject DTMF',
+                  'reuse codecs',
+                ],
                 sdp
               };
               const response = await this.answer(opts);
@@ -672,7 +679,21 @@ class CallSession extends Emitter {
         }
       }
     } catch (err) {
-      if ('abandonded' !== err.message) this.logger.error({err}, `Error setting up outbound call to: ${uris}`);
+      if ('abandoned' !== err.message) this.logger.error({err}, `Error setting up outbound call to: ${uris}`);
+
+      /* the call count was incremented on 'init' but we bailed out before connecting (e.g. caller
+         hung up before the B leg answered), so we must decrement it here - no failure response is
+         sent on this path, so the res.once('end') safety net in middleware would not fire.
+         nudgeCallCounts is a no-op if the count was already decremented elsewhere. */
+      const {writeCallCount, writeCallCountSP, writeCallCountApp} = this.req.srf.locals;
+      nudgeCallCounts(this.req, 'failure', {
+        service_provider_sid: this.service_provider_sid,
+        account_sid: this.account_sid,
+        application_sid: this.application_sid,
+        callId: this.req.locals.callId
+      }, this.decrKey, {writeCallCountSP, writeCallCount, writeCallCountApp})
+        .catch((err) => this.logger.error(err, 'Error decrementing call counts'));
+
       this.emit('failed');
       this.srf.endSession(this.req);
       this.rtpEngineResource.destroy();
@@ -737,10 +758,13 @@ class CallSession extends Emitter {
         if (this.req.locals.cdr) {
           const now = Date.now();
           const day = new Date();
-          const recordAllCalls = this.req.locals.record_all_calls;
+          // eslint-disable-next-line max-len
+          const {hasRecording = false} = await this.retrieveHash(`call:${this.account_sid}:${this.req.locals.cdr.call_sid}`) ?? {};
+          const recordAllCalls = this.req.locals.record_all_calls || hasRecording;
+          const record_format = this.req.locals.account.record_format || 'mp3';
           let recording_url = `/Accounts/${this.account_sid}/RecentCalls/${this.req.locals.cdr.call_sid}/record`;
           recording_url += `/${day.getFullYear()}/${(day.getMonth() + 1).toString().padStart(2, '0')}`;
-          recording_url += `/${day.getDate().toString().padStart(2, '0')}/${recordAllCalls}`;
+          recording_url += `/${day.getDate().toString().padStart(2, '0')}/${record_format}`;
           this.writeCdrs({...this.req.locals.cdr,
             terminated_at: now,
             termination_reason: dlg.type === 'uas' ? 'caller hungup' : 'called party hungup',
@@ -896,8 +920,14 @@ Duration=${payload.duration} `
         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' }})};
+      answerMedia = { ...answerMedia, ...((process.env.JAMBONES_CODEC_OFFER_WITH_ORDER ||
+        process.env.JAMBONES_CODEC_TRANSCODE) && dlg.type === 'uac' &&
+        { codec: {
+          ...(process.env.JAMBONES_CODEC_OFFER_WITH_ORDER &&
+            { offer: process.env.JAMBONES_CODEC_OFFER_WITH_ORDER.split(','), strip: 'all' }),
+          ...(process.env.JAMBONES_CODEC_TRANSCODE &&
+            { transcode: process.env.JAMBONES_CODEC_TRANSCODE.split(',') })
+        }})};
       let opts = {
         ...this.rtpEngineOpts.common,
         ...offerMedia,
@@ -1222,7 +1252,6 @@ Duration=${payload.duration} `
         ...(req.has('X-Retain-Call-Sid') && {'X-Retain-Call-Sid': req.get('X-Retain-Call-Sid')}),
         ...(req.has('X-Account-Sid') && {'X-Account-Sid': req.get('X-Account-Sid')})
       };
-
       const uac = await this.srf.createUAC(referTo.uri, {localSdp: dlg.local.sdp, headers});
       this.uas = uac;
       uac.type = 'uas';
@@ -1231,19 +1260,23 @@ Duration=${payload.duration} `
       uac.on('info', this._onInfo.bind(this, uac));
       uac.on('modify', this._onReinvite.bind(this, uac));
       uac.on('refer', this._onFeatureServerTransfer.bind(this, uac));
-      uac.on('destroy', () => {
+      uac.on('destroy', async() => {
         this.logger.info('call ended with normal termination');
         this.rtpEngineResource.destroy();
         this.activeCallIds.delete(this.req.get('Call-ID'));
         if (this.activeCallIds.size === 0) this.idleEmitter.emit('idle');
         uac.other.destroy();
+        this._stopRecording();
         if (this.req.locals.cdr) {
           const now = Date.now();
           const day = new Date();
-          const recordAllCalls = this.req.locals.record_all_calls;
+          // eslint-disable-next-line max-len
+          const {hasRecording = false} = await this.retrieveHash(`call:${this.account_sid}:${this.req.locals.cdr.call_sid}`) ?? {} ;
+          const recordAllCalls = this.req.locals.record_all_calls || hasRecording;
+          const record_format = this.req.locals.account.record_format || 'mp3';
           let recording_url = `/Accounts/${this.account_sid}/RecentCalls/${this.req.locals.cdr.call_sid}/record`;
           recording_url += `/${day.getFullYear()}/${(day.getMonth() + 1).toString().padStart(2, '0')}`;
-          recording_url += `/${day.getDate().toString().padStart(2, '0')}/${recordAllCalls}`;
+          recording_url += `/${day.getDate().toString().padStart(2, '0')}/${record_format}`;
           this.writeCdrs({...this.req.locals.cdr,
             terminated_at: now,
             termination_reason: 'caller hungup',

lib/utils.js

@@ -5,7 +5,8 @@ const CIDRMatcher = require('cidr-matcher');
 const dns = require('dns');
 const sdpTransform = require('sdp-transform');
 
-function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams) {
+// eslint-disable-next-line max-len
+function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams, remove_ice = false, dtls_off = false) {
   const from = req.getParsedHeader('from');
   const rtpCopy = JSON.parse(JSON.stringify(rtpCharacteristics));
   const srtpCopy = JSON.parse(JSON.stringify(srtpCharacteristics));
@@ -16,8 +17,16 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams
   }
 
   const srtpOpts = teams ? srtpCopy['teams'] : srtpCopy['default'];
-  const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCopy;
-  const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCopy;
+
+  if (remove_ice) {
+    srtpOpts.ICE = 'remove';
+  }
+  if (dtls_off) {
+    srtpOpts.DTLS = 'off';
+  }
+
+  const dstOpts = JSON.parse(JSON.stringify(dstIsUsingSrtp ? srtpOpts : rtpCopy));
+  const srcOpts = JSON.parse(JSON.stringify(srcIsUsingSrtp ? srtpOpts : rtpCopy));
 
   /** Allow feature server to send DTMF to the call excepts call from/to teams */
   if (!teams) {
@@ -41,6 +50,20 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams
     'replace': ['origin', 'session-connection'],
     'record call': process.env.JAMBONES_RECORD_ALL_CALLS ? 'yes' : 'no'
   };
+
+  const codec = {
+    accept: ['PCMU', 'PCMA', 'telephone-event'],
+    ...(process.env.JAMBONES_CODEC_OFFER_WITH_ORDER &&
+      {
+        offer: process.env.JAMBONES_CODEC_OFFER_WITH_ORDER.split(','),
+        strip: 'all'
+      }),
+    ...(process.env.JAMBONES_CODEC_TRANSCODE &&
+      {
+        transcode: process.env.JAMBONES_CODEC_TRANSCODE.split(',')
+      }),
+  };
+
   return {
     common,
     uas: {
@@ -51,8 +74,7 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams
       tag: null,
       mediaOpts: {
         ...dstOpts,
-        ...(process.env.JAMBONES_CODEC_OFFER_WITH_ORDER &&
-          { codec: { offer: process.env.JAMBONES_CODEC_OFFER_WITH_ORDER.split(','), strip: 'all' } }),
+        codec,
       }
     }
   };
@@ -350,54 +372,6 @@ const removeVideoSdp = (sdp) => {
   return sdpTransform.write(parsedSdp);
 };
 
-const determineAnswerCodec = (farEndSdp, featureServerSdp, logger) => {
-  try {
-    // Parse both SDPs
-    const farEndParsed = sdpTransform.parse(farEndSdp);
-    const fsParsed = sdpTransform.parse(featureServerSdp);
-
-    // Get negotiated codec from far end (first codec in answer)
-    const negotiatedCodec = farEndParsed.media[0].rtp[0].codec;
-
-    // Get all codecs offered by feature server
-    const fsCodecs = fsParsed.media[0].rtp.map((r) => r.codec);
-
-    logger.debug({negotiatedCodec, fsCodecs}, 'determineAnswerCodec: analyzing codec negotiation');
-
-    // If far end negotiated G.711 (PCMU/PCMA) AND it was in the FS offer, pass it through
-    if (['PCMU', 'PCMA'].includes(negotiatedCodec) && fsCodecs.includes(negotiatedCodec)) {
-      logger.info({negotiatedCodec}, 'G.711 codec passthrough - no transcoding needed');
-      return {
-        codec: negotiatedCodec,
-        needsTranscoding: false
-      };
-    }
-
-    // Otherwise, we need to transcode to first G.711 codec in FS offer
-    const firstG711 = fsCodecs.find((c) => ['PCMU', 'PCMA'].includes(c));
-    if (firstG711) {
-      logger.info({negotiatedCodec, transcodeTarget: firstG711}, 'Transcoding required to G.711');
-      return {
-        codec: firstG711,
-        needsTranscoding: true
-      };
-    }
-
-    // Fallback: use PCMU
-    logger.info({negotiatedCodec}, 'No G.711 in FS offer, defaulting to PCMU');
-    return {
-      codec: 'PCMU',
-      needsTranscoding: true
-    };
-  } catch (err) {
-    logger.error({err}, 'Error determining answer codec, defaulting to PCMU');
-    return {
-      codec: 'PCMU',
-      needsTranscoding: true
-    };
-  }
-};
-
 module.exports = {
   makeRtpEngineOpts,
   selectHostPort,
@@ -414,5 +388,4 @@ module.exports = {
   makePartnerFullMediaReleaseKey,
   isValidDomainOrIP,
   removeVideoSdp,
-  determineAnswerCodec
 };

package.json

@@ -1,9 +1,9 @@
 {
   "name": "sbc-outbound",
-  "version": "0.9.5",
+  "version": "0.9.11",
   "main": "app.js",
   "engines": {
-    "node": ">= 18.0.0"
+    "node": ">= 20.0.0"
   },
   "keywords": [
     "sip",
@@ -31,7 +31,7 @@
     "@jambonz/db-helpers": "^0.9.18",
     "@jambonz/http-health-check": "^0.0.1",
     "@jambonz/mw-registrar": "0.2.7",
-    "@jambonz/realtimedb-helpers": "^0.8.13",
+    "@jambonz/realtimedb-helpers": "^0.8.21",
     "@jambonz/rtpengine-utils": "^0.4.4",
     "@jambonz/siprec-client-utils": "^0.2.10",
     "@jambonz/stats-collector": "^0.1.10",
@@ -39,7 +39,7 @@
     "cidr-matcher": "^2.1.1",
     "debug": "^4.4.3",
     "drachtio-fn-b2b-sugar": "^0.2.1",
-    "drachtio-srf": "^5.0.12",
+    "drachtio-srf": "^5.0.21",
     "express": "^4.21.2",
     "pino": "^10.1.0",
     "sdp-transform": "^2.15.0"
@@ -48,7 +48,7 @@
     "bent": "^7.3.12",
     "eslint": "^9.17.0",
     "eslint-plugin-promise": "^7.2.1",
-    "nyc": "^15.1.0",
-    "tape": "^5.7.5"
+    "nyc": "^17.1.0",
+    "tape": "^5.9.0"
   }
 }