bump version to 0.7.6

Snyk has created this PR to upgrade drachtio-srf from 4.5.0 to 4.5.1.

See this package in npm:
https://www.npmjs.com/package/drachtio-srf

See this project in Snyk:
https://app.snyk.io/org/davehorton/project/282b1881-3e13-4fad-85bd-3b1c662c2134?utm_source=github&utm_medium=referral&page=upgrade-pr

Dockerfile

@@ -1,10 +1,23 @@
-FROM node:17-slim
+FROM --platform=linux/amd64 node:18.6.0-alpine as base
+
+RUN apk --update --no-cache add --virtual .builds-deps build-base python3
+
 WORKDIR /opt/app/
-COPY package.json ./
-RUN npm install
-RUN npm prune
-COPY . /opt/app
+
+FROM base as build
+
+COPY package.json package-lock.json ./
+
+RUN npm ci
+
+COPY . .
+
+FROM base
+
+COPY --from=build /opt/app /opt/app/
+
 ARG NODE_ENV
+
 ENV NODE_ENV $NODE_ENV
 
-CMD [ "npm", "start" ]
+CMD [ "node", "app.js" ]

lib/call-session.js

@@ -1,4 +1,6 @@
 const Emitter = require('events');
+const sdpTransform = require('sdp-transform');
+const SrsClient = require('@jambonz/siprec-client-utils');
 const {makeRtpEngineOpts, makeCallCountKey} = require('./utils');
 const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
 const {SipError, stringifyUri, parseUri} = require('drachtio-srf');
@@ -48,6 +50,15 @@ const initCdr = (srf, req) => {
   };
 };
 
+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}`);
+  } catch (err) {}
+  return opts;
+};
+
 class CallSession extends Emitter {
   constructor(logger, req, res) {
     super();
@@ -68,6 +79,8 @@ class CallSession extends Emitter {
     this.performLcr = performLcr;
     this.lookupCarrierBySid = lookupCarrierBySid;
     this.lookupSipGatewaysByCarrier = lookupSipGatewaysByCarrier;
+
+    this._mediaReleased = false;
   }
 
   get account_sid() {
@@ -78,6 +91,10 @@ class CallSession extends Emitter {
     return this.srf.locals.privateSipAddress;
   }
 
+  get isMediaReleased() {
+    return this._mediaReleased;
+  }
+
   async connect() {
     const teams = this.teams = this.req.locals.target === 'teams';
     const engine = this.srf.locals.getRtpEngine();
@@ -94,8 +111,12 @@ class CallSession extends Emitter {
       unblockMedia,
       blockDTMF,
       unblockDTMF,
+      playDTMF,
       subscribeDTMF,
-      unsubscribeDTMF
+      unsubscribeDTMF,
+      subscribeRequest,
+      subscribeAnswer,
+      unsubscribe
     } = engine;
     const {createHash, retrieveHash} = this.srf.locals.realtimeDbHelpers;
     this.offer = offer;
@@ -105,8 +126,12 @@ class CallSession extends Emitter {
     this.unblockMedia = unblockMedia;
     this.blockDTMF = blockDTMF;
     this.unblockDTMF = unblockDTMF;
+    this.playDTMF = playDTMF;
     this.subscribeDTMF = subscribeDTMF;
     this.unsubscribeDTMF = unsubscribeDTMF;
+    this.subscribeRequest = subscribeRequest;
+    this.subscribeAnswer = subscribeAnswer;
+    this.unsubscribe = unsubscribe;
 
     this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, teams);
     this.rtpEngineResource = {destroy: this.del.bind(null, this.rtpEngineOpts.common)};
@@ -221,13 +246,13 @@ class CallSession extends Emitter {
       }
 
       // rtpengine 'offer'
-      const opts = {
+      const opts = updateRtpEngineFlags(this.req.body, {
         ...this.rtpEngineOpts.common,
         ...this.rtpEngineOpts.uac.mediaOpts,
         'from-tag': this.rtpEngineOpts.uas.tag,
         direction:  ['private', 'public'],
         sdp: this.req.body
-      };
+      });
       const response = await this.offer(opts);
       debug(`response from rtpengine to offer ${JSON.stringify(response)}`);
       this.logger.debug({offer: opts, response}, 'initial offer to rtpengine');
@@ -552,6 +577,8 @@ Duration=${payload.duration} `
       if (reason && dlg.type === 'uas' && ['release-media', 'anchor-media'].includes(reason)) {
         this.logger.info(`got a reinvite from FS to ${reason}`);
         sdp = dlg.other.remote.sdp;
+        answerMedia.flags = ['asymmetric', 'port latching'];
+        this._mediaReleased = 'release-media' === reason;
       }
       else {
         sdp = await dlg.other.modify(response.sdp);
@@ -578,6 +605,7 @@ Duration=${payload.duration} `
 
   async _onInfo(dlg, req, res) {
     try {
+      const contentType = req.get('Content-Type');
       if (dlg.type === 'uas' && req.has('X-Reason')) {
         const toTag = this.rtpEngineOpts.uac.tag;
         const reason = req.get('X-Reason');
@@ -596,6 +624,123 @@ Duration=${payload.duration} `
           const response = Promise.all([this.unblockMedia(opts), this.unblockDTMF(opts)]);
           this.logger.info({response}, `_onInfo: response to rtpengine command for ${reason}`);
         }
+        else if (reason.includes('CallRecording')) {
+          let succeeded = false;
+          if (reason === 'startCallRecording') {
+            const from = this.req.getParsedHeader('From');
+            const to = this.req.getParsedHeader('To');
+            const aorFrom = from.uri;
+            const aorTo = to.uri;
+            this.logger.info({to, from}, 'startCallRecording request for an outbound call');
+
+            const srsUrl = req.get('X-Srs-Url');
+            const srsRecordingId = req.get('X-Srs-Recording-ID');
+            const callSid = req.get('X-Call-Sid');
+            const accountSid = req.get('X-Account-Sid');
+            const applicationSid = req.get('X-Application-Sid');
+            if (this.srsClient) {
+              res.send(400);
+              this.logger.info('discarding duplicate startCallRecording request for a call');
+              return;
+            }
+            if (!srsUrl) {
+              this.logger.info('startCallRecording request is missing X-Srs-Url header');
+              res.send(400);
+              return;
+            }
+            this.srsClient = new SrsClient(this.logger, {
+              srf: dlg.srf,
+              direction: 'outbound',
+              originalInvite: this.req,
+              callingNumber: this.req.callingNumber,
+              calledNumber: this.req.calledNumber,
+              srsUrl,
+              srsRecordingId,
+              callSid,
+              accountSid,
+              applicationSid,
+              rtpEngineOpts: this.rtpEngineOpts,
+              toTag,
+              aorFrom,
+              aorTo,
+              subscribeRequest: this.subscribeRequest,
+              subscribeAnswer: this.subscribeAnswer,
+              del: this.del,
+              blockMedia: this.blockMedia,
+              unblockMedia: this.unblockMedia,
+              unsubscribe: this.unsubscribe
+            });
+            try {
+              succeeded = await this.srsClient.start();
+            } catch (err) {
+              this.logger.error({err}, 'Error starting SipRec call recording');
+            }
+          }
+          else if (reason === 'stopCallRecording') {
+            if (!this.srsClient) {
+              res.send(400);
+              this.logger.info('discarding stopCallRecording request because we are not recording');
+              return;
+            }
+            try {
+              succeeded = await this.srsClient.stop();
+            } catch (err) {
+              this.logger.error({err}, 'Error stopping SipRec call recording');
+            }
+            this.srsClient = null;
+          }
+          else if (reason === 'pauseCallRecording') {
+            if (!this.srsClient || this.srsClient.paused) {
+              this.logger.info('discarding invalid pauseCallRecording request');
+              res.send(400);
+              return;
+            }
+            succeeded = await this.srsClient.pause();
+          }
+          else if (reason === 'resumeCallRecording') {
+            if (!this.srsClient || !this.srsClient.paused) {
+              res.send(400);
+              this.logger.info('discarding invalid resumeCallRecording request');
+              return;
+            }
+            succeeded = await this.srsClient.resume();
+          }
+          res.send(succeeded ? 200 : 503);
+        }
+      }
+      else if (dlg.type === 'uac' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
+        const arr = /Signal=\s*([1-9#*])/.exec(req.body);
+        if (!arr) {
+          this.logger.info({body: req.body}, '_onInfo: invalid INFO dtmf request');
+          throw new Error(`_onInfo: no dtmf in body for ${contentType}`);
+        }
+        const code = arr[1];
+        const arr2 = /Duration=\s*(\d+)/.exec(req.body);
+        const duration = arr2 ? arr2[1] : 250;
+
+        if (this.isMediaReleased) {
+          /* just relay on to the feature server */
+          this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, relaying to feature server');
+          this._onDTMF(dlg.other, {event: code, duration})
+            .catch((err) => this.logger.info({err}, 'Error relaying DTMF to feature server'));
+          res.send(200);
+        }
+        else {
+          /* else convert SIP INFO to RFC 2833 telephony events */
+          this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, converting to RFC 2833');
+          const opts = {
+            ...this.rtpEngineOpts.common,
+            'from-tag': this.rtpEngineOpts.uac.tag,
+            code,
+            duration
+          };
+          const response = await this.playDTMF(opts);
+          if ('ok' !== response.result) {
+            this.logger.info({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
+            throw new Error('rtpengine failed: answer');
+          }
+          res.send(200);
+        }
       }
       else {
         const response = await dlg.other.request({

lib/utils.js

@@ -7,6 +7,12 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
   const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
   const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
   const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCharacteristics;
+
+  /* webrtc clients (e.g. sipjs) send DMTF via SIP INFO */
+  if ((srcIsUsingSrtp || dstIsUsingSrtp) && !teams) {
+    dstOpts.flags.push('inject DTMF');
+    srcOpts.flags.push('inject DTMF');
+  }
   const common = {
     'call-id': req.get('Call-ID'),
     'replace': ['origin', 'session-connection']

package.json

@@ -1,6 +1,6 @@
 {
   "name": "sbc-outbound",
-  "version": "v0.7.5",
+  "version": "v0.7.6",
   "main": "app.js",
   "engines": {
     "node": ">= 12.0.0"
@@ -30,21 +30,24 @@
     "@jambonz/db-helpers": "^0.6.18",
     "@jambonz/http-health-check": "^0.0.1",
     "@jambonz/mw-registrar": "0.2.2",
-    "@jambonz/realtimedb-helpers": "^0.4.24",
+    "@jambonz/realtimedb-helpers": "^0.4.29",
     "@jambonz/rtpengine-utils": "^0.3.1",
+    "@jambonz/siprec-client-utils": "^0.1.4",
     "@jambonz/stats-collector": "^0.1.6",
-    "@jambonz/time-series": "^0.1.9",
+    "@jambonz/time-series": "^0.1.12",
     "cidr-matcher": "^2.1.1",
-    "debug": "^4.3.3",
+    "debug": "^4.3.4",
     "drachtio-fn-b2b-sugar": "^0.0.12",
-    "drachtio-srf": "^4.4.59",
-    "husky": "^7.0.4",
-    "pino": "^7.4.1"
+    "drachtio-srf": "^4.5.1",
+    "express": "^4.18.1",
+    "pino": "^7.11.0",
+    "sdp-transform": "^2.14.1"
   },
   "devDependencies": {
+    "bent": "^7.3.12",
     "eslint": "^7.32.0",
-    "eslint-plugin-promise": "^5.1.1",
+    "eslint-plugin-promise": "^5.2.0",
     "nyc": "^15.1.0",
-    "tape": "^5.3.2"
+    "tape": "^5.5.3"
   }
 }

test/registrar.env

@@ -5,3 +5,7 @@ DRACHTIO_SECRET=cymru
 JAMBONES_REDIS_HOST=172.39.0.11
 JAMBONES_REDIS_PORT=6379
 JAMBONES_LOGLEVEL=info
+JAMBONES_MYSQL_HOST=172.39.0.2
+JAMBONES_MYSQL_USER=jambones_test
+JAMBONES_MYSQL_PASSWORD=jambones_test
+JAMBONES_MYSQL_DATABASE=jambones_test