after challenge register with creds should stay tls if it started that way (#137)

* only send Sips if it's enabled in outbound gateway configuration

* update license

LICENSE

@@ -1,6 +1,6 @@
 MIT License
 
-Copyright (c) 2021 Drachtio Communications Services, LLC
+Copyright (c) 2018-2024 FirstFive8, Inc.
 
 Permission is hereby granted, free of charge, to any person obtaining a copy
 of this software and associated documentation files (the "Software"), to deal

app.js

@@ -14,30 +14,6 @@ assert.ok(process.env.DRACHTIO_SECRET, 'missing DRACHTIO_SECRET env var');
 assert.ok(process.env.JAMBONES_NETWORK_CIDR || process.env.K8S, 'missing JAMBONES_NETWORK_CIDR env var');
 assert.ok(process.env.JAMBONES_TIME_SERIES_HOST, 'missing JAMBONES_TIME_SERIES_HOST env var');
 
-const JAMBONES_REDIS_SENTINELS = process.env.JAMBONES_REDIS_SENTINELS ? {
-  sentinels: process.env.JAMBONES_REDIS_SENTINELS.split(',').map((sentinel) => {
-    let host, port = 26379;
-    if (sentinel.includes(':')) {
-      const arr = sentinel.split(':');
-      host = arr[0];
-      port = parseInt(arr[1], 10);
-    } else {
-      host = sentinel;
-    }
-    return {host, port};
-  }),
-  name: process.env.JAMBONES_REDIS_SENTINEL_MASTER_NAME,
-  ...(process.env.JAMBONES_REDIS_SENTINEL_PASSWORD && {
-    password: process.env.JAMBONES_REDIS_SENTINEL_PASSWORD
-  }),
-  ...(process.env.JAMBONES_REDIS_SENTINEL_USERNAME && {
-    username: process.env.JAMBONES_REDIS_SENTINEL_USERNAME
-  }),
-  ...(process.env.JAMBONES_REDIS_SENTINEL_SENTINAL_PASSWORD && {
-    sentinelPassword: process.env.JAMBONES_REDIS_SENTINEL_SENTINAL_PASSWORD
-  }),
-} : null;
-
 const Srf = require('drachtio-srf');
 const srf = new Srf('sbc-outbound');
 const CIDRMatcher = require('cidr-matcher');
@@ -91,10 +67,7 @@ const {
   decrKey,
   retrieveSet,
   isMemberOfSet,
-} = require('@jambonz/realtimedb-helpers')(JAMBONES_REDIS_SENTINELS || {
-  host: process.env.JAMBONES_REDIS_HOST,
-  port: process.env.JAMBONES_REDIS_PORT || 6379
-}, logger);
+} = require('@jambonz/realtimedb-helpers')({}, logger);
 
 const activeCallIds = new Map();
 const Emitter = require('events');
@@ -124,6 +97,7 @@ srf.locals = {...srf.locals,
     lookupCarrierByAccountLcr
   },
   realtimeDbHelpers: {
+    client: redisClient,
     createHash,
     retrieveHash,
     incrKey,

lib/call-session.js

@@ -1,17 +1,18 @@
 const Emitter = require('events');
 const sdpTransform = require('sdp-transform');
 const SrsClient = require('@jambonz/siprec-client-utils');
-const {makeRtpEngineOpts, nudgeCallCounts} = require('./utils');
+const {makeRtpEngineOpts, nudgeCallCounts, isPrivateVoipNetwork, isBlackListedSipGateway} = require('./utils');
 const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
 const {SipError, stringifyUri, parseUri} = require('drachtio-srf');
 const debug = require('debug')('jambonz:sbc-outbound');
 
 const makeInviteInProgressKey = (callid) => `sbc-out-iip${callid}`;
+const IMMUTABLE_HEADERS = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
 /**
  * this is to make sure the outgoing From has the number in the incoming From
  * and not the incoming PAI
  */
-const createBLegFromHeader = (req, teams) => {
+const createBLegFromHeader = (req, teams, register_from_domain = null) => {
   const from = req.getParsedHeader('From');
   const uri = parseUri(from.uri);
   let user = uri.user || 'anonymous';
@@ -22,6 +23,11 @@ const createBLegFromHeader = (req, teams) => {
   else if (req.has('X-Preferred-From-User') || req.has('X-Preferred-From-Host')) {
     user = req.get('X-Preferred-From-User') || user;
     host = req.get('X-Preferred-From-Host') || host;
+  } else if (register_from_domain) {
+    host = register_from_domain;
+  }
+  if (from.name) {
+    return `${from.name} <sip:${user}@${host}>`;
   }
   return `sip:${user}@${host}`;
 };
@@ -97,6 +103,7 @@ class CallSession extends Emitter {
     this.lookupCarrierByAccountLcr = lookupCarrierByAccountLcr;
 
     this._mediaReleased = false;
+    this.recordingNoAnswerTimeout = (process.env.JAMBONES_RECORDING_NO_ANSWER_TIMEOUT || 2) * 1000;
   }
 
   get service_provider_sid() {
@@ -161,7 +168,7 @@ class CallSession extends Emitter {
       subscribeAnswer,
       unsubscribe
     } = engine;
-    const {createHash, retrieveHash} = this.srf.locals.realtimeDbHelpers;
+    const {client, createHash, retrieveHash} = this.srf.locals.realtimeDbHelpers;
     this.offer = offer;
     this.answer = answer;
     this.del = del;
@@ -179,8 +186,9 @@ class CallSession extends Emitter {
 
     this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, false, teams);
     this.rtpEngineResource = {destroy: this.del.bind(null, this.rtpEngineOpts.common)};
-    let proxy, uris;
+    let proxy, uris = [];
     const mapGateways = new Map();
+    let encryptedMedia = false;
 
     try {
       // determine where to send the call
@@ -204,21 +212,37 @@ class CallSession extends Emitter {
           destUri = stringifyUri(uri);
           this.logger.info(`overriding destination user with ${dest}, so final uri is ${destUri}`);
         }
-        uris = [destUri];
+        uris = [{
+          private_network: contact.includes('transport=ws') ? false : await isPrivateVoipNetwork(destUri),
+          uri: destUri
+        }];
         if (!contact.includes('transport=ws')) {
           proxy = this.req.locals.registration.proxy;
         }
       }
       else if (this.req.locals.target === 'forward') {
-        uris = [this.req.uri];
+        uris = [{
+          private_network: await isPrivateVoipNetwork(this.req.uri),
+          uri: this.req.uri
+        }];
+        if (this.req.has('X-SIP-Proxy')) {
+          proxy = this.req.get('X-SIP-Proxy');
+          if (!proxy.startsWith('sip:') && !proxy.startsWith('sips:')) proxy = `sip:${proxy}`;
+        }
       }
       else if (teams) {
         const vmailParam = 'opaque=app:voicemail';
         proxy = `sip:${this.req.calledNumber}@sip.pstnhub.microsoft.com:5061;transport=tls`;
         if (this.req.uri.includes(vmailParam)) {
-          uris = [`sip:${this.req.calledNumber}@sip.pstnhub.microsoft.com;${vmailParam}`];
+          uris = [{
+            private_network: false,
+            uri: `sip:${this.req.calledNumber}@sip.pstnhub.microsoft.com;${vmailParam}`
+          }];
         }
-        else uris = [`sip:${this.req.calledNumber}@sip.pstnhub.microsoft.com`];
+        else uris = [{
+          private_network: false,
+          uri: `sip:${this.req.calledNumber}@sip.pstnhub.microsoft.com`
+        }];
         headers = {
           ...headers,
           Contact: `sip:${this.req.calledNumber}@${this.req.get('X-MS-Teams-Tenant-FQDN')}:5061;transport=tls`
@@ -257,7 +281,13 @@ class CallSession extends Emitter {
           }
           const vc = await this.lookupCarrierBySid(voip_carrier_sid);
           const gateways = await this.lookupSipGatewaysByCarrier(voip_carrier_sid);
-          const gws = (gateways || [])
+          const goodGateways = [];
+          for (const g of gateways) {
+            if (!await isBlackListedSipGateway(client, this.logger, g.sip_gateway_sid)) {
+              goodGateways.push(g);
+            }
+          }
+          const gws = (goodGateways || [])
             .filter((gw) => gw.outbound);
           if (gws.length) {
             uris = [];
@@ -270,13 +300,17 @@ class CallSession extends Emitter {
               const hostport = !o.port || 5060 === o.port ? o.ipv4 : `${o.ipv4}:${o.port}`;
               const prependPlus = vc.e164_leading_plus && !this.req.calledNumber.startsWith('0') ? '+' : '';
               const transport = `transport=${protocol}`;
-              const scheme = protocol === 'tls' ? 'sips' : 'sip';
+              const useSipsScheme = protocol === 'tls' &&
+                !process.env.JAMBONES_USE_BEST_EFFORT_TLS &&
+                o.use_sips_scheme;
+              const scheme = useSipsScheme ? 'sips' : 'sip';
               const u = `${scheme}:${prefix}${prependPlus}${calledNumber}@${hostport};${transport}`;
               const obj = {
                 name: vc.name,
                 diversion: vc.diversion,
                 hostport,
-                protocol: o.protocol
+                protocol: o.protocol,
+                register_from_domain: vc.register_from_domain
               };
               if (vc.register_username && vc.register_password) {
                 obj.auth = {
@@ -295,12 +329,21 @@ class CallSession extends Emitter {
                  */
                 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);
+                encryptedMedia = true;
               }
             });
+            // Check private network for each gw
+            uris = await Promise.all(uris.map(async(u) => {
+              return {
+                private_network: await isPrivateVoipNetwork(u),
+                uri: u
+              };
+            }));
             this.logger.debug({uris, voip_carrier_sid}, 'selected outbound gateways for requested carrier');
           }
           else {
             this.logger.info({voip_carrier_sid}, 'no outbound gateways found for requested carrier');
+            this.res.send(603);
           }
         } catch (err) {
           debug(err);
@@ -311,15 +354,22 @@ class CallSession extends Emitter {
         debug(`sending call to PSTN ${uris}`);
       }
 
+      // private_network should be called at last
+      uris = uris.sort((a, b) => a.private_network - b.private_network);
+      const toPrivate = uris.some((u) => u.private_network === true);
+      const toPublic = uris.some((u) => u.private_network === false);
+      let isOfferUpdatedToPrivate = toPrivate && !toPublic;
+
+
       // rtpengine 'offer'
       const opts = updateRtpEngineFlags(this.req.body, {
         ...this.rtpEngineOpts.common,
         ...this.rtpEngineOpts.uac.mediaOpts,
         'from-tag': this.rtpEngineOpts.uas.tag,
-        direction:  ['private', 'public'],
+        direction:  ['private', toPublic ? 'public' : 'private'],
         sdp: this.req.body
       });
-      const response = await this.offer(opts);
+      let response = await this.offer(opts);
       debug(`response from rtpengine to offer ${JSON.stringify(response)}`);
       this.logger.debug({offer: opts, response}, 'initial offer to rtpengine');
       if ('ok' !== response.result) {
@@ -334,7 +384,17 @@ class CallSession extends Emitter {
       let earlyMedia = false;
       while (uris.length) {
         let hdrs = { ...headers};
-        const uri = uris.shift();
+        const {private_network, uri} = uris.shift();
+        if (private_network && !isOfferUpdatedToPrivate) {
+          // Cannot make call to all public Uris, now come to talk with private network Uris
+          this.rtpEngineResource.destroy()
+            .catch((err) => this.logger.info({err}, 'Error destroying rtpe to re-connect to private network'));
+          response = await this.offer({
+            ...opts,
+            direction: ['private', 'private']
+          });
+          isOfferUpdatedToPrivate = true;
+        }
         const gw = mapGateways.get(uri);
         const passFailure = 0 === uris.length;  // only a single target
         if (0 === uris.length) {
@@ -367,6 +427,12 @@ class CallSession extends Emitter {
               'Diversion': div
             };
           }
+          if (gw.register_from_domain) {
+            hdrs = {
+              ...hdrs,
+              'From': createBLegFromHeader(this.req, teams, gw.register_from_domain)
+            };
+          }
         }
         else this.logger.info(`sending INVITE to ${uri} via proxy ${proxy})`);
         try {
@@ -389,6 +455,7 @@ class CallSession extends Emitter {
               '-X-Preferred-From-Host',
               '-X-Jambonz-FS-UUID',
               '-X-Voip-Carrier-Sid',
+              '-X-SIP-Proxy'
             ],
             proxyResponseHeaders: [
               'all',
@@ -406,7 +473,7 @@ class CallSession extends Emitter {
                 ...this.rtpEngineOpts.uas.mediaOpts,
                 'from-tag': this.rtpEngineOpts.uas.tag,
                 'to-tag': this.rtpEngineOpts.uac.tag,
-                flags: ['single codec'],
+                flags: ['single codec', 'inject DTMF'],
                 sdp
               };
               const response = await this.answer(opts);
@@ -468,6 +535,7 @@ class CallSession extends Emitter {
           // these are all final failure scenarios
           if (uris.length === 0 ||          // exhausted all targets
             earlyMedia ||                   // failure after early media
+            encryptedMedia ||               // cant crank back when using encrypted media as keys have been exchanged
             !(err instanceof SipError) ||   // unexpected error
             err.status === 487) {           // caller hung up
 
@@ -521,13 +589,18 @@ class CallSession extends Emitter {
     this.uas = uas;
     this.uac = uac;
     [uas, uac].forEach((dlg) => {
-      dlg.on('destroy', async() => {
+      dlg.on('destroy', async(bye) => {
         const other = dlg.other;
         this.rtpEngineResource.destroy();
         this.activeCallIds.delete(this.req.get('Call-ID'));
         this.unsubscribeForDTMF();
         //this.unsubscribeDTMF(this.logger, this.req.get('Call-ID'), this.rtpEngineOpts.uac.tag);
         try {
+          const headers = {};
+          Object.keys(bye.headers).forEach((h) => {
+            if (!IMMUTABLE_HEADERS.includes(h)) headers[h] = bye.headers[h];
+          });
+          await other.destroy({headers});
           await other.destroy();
         } catch (err) {}
 
@@ -593,6 +666,22 @@ class CallSession extends Emitter {
     forwardInDialogRequests(uac,  ['notify', 'options', 'message']);
   }
 
+  _startRecordingNoAnswerTimer(res) {
+    this._clearRecordingNoAnswerTimer();
+    this.recordingNoAnswerTimer = setTimeout(() => {
+      this.logger.info('No response from SipRec server, return error to feature server');
+      this.isRecordingNoAnswerResponded = true;
+      res.send(400);
+    }, this.recordingNoAnswerTimeout);
+  }
+
+  _clearRecordingNoAnswerTimer() {
+    if (this.recordingNoAnswerTimer) {
+      clearTimeout(this.recordingNoAnswerTimer);
+      this.recordingNoAnswerTimer = null;
+    }
+  }
+
   _stopRecording() {
     if (this.srsClients.length) {
       this.srsClients.forEach((c) => c.stop());
@@ -699,6 +788,8 @@ Duration=${payload.duration} `
         sdp
       };
       response = await this.answer(opts);
+      /* now remove asymeetric as B party (looking at you Genesys ring group) may need port re-learning on invites  */
+      answerMedia.flags = answerMedia.flags.filter((f) => f !== 'asymmetric');
       if ('ok' !== response.result) {
         res.send(488);
         throw new Error(`_onReinvite: rtpengine failed: ${JSON.stringify(response)}`);
@@ -706,6 +797,7 @@ Duration=${payload.duration} `
       this.logger.debug({opts, sdp: response.sdp}, 'CallSession:_onReinvite: (answer) sending back upstream');
       res.send(200, {body: response.sdp});
     } catch (err) {
+      res.send(err.status || 500);
       this.logger.error(err, 'Error handling reinvite');
     }
   }
@@ -733,12 +825,12 @@ Duration=${payload.duration} `
         }
         else if (reason.includes('CallRecording')) {
           let succeeded = false;
+          const headers = contentType === 'application/json' && req.body ? JSON.parse(req.body) : {};
           if (reason === 'startCallRecording') {
             const from = this.req.getParsedHeader('From');
             const to = this.req.getParsedHeader('To');
             const aorFrom = from.uri;
             const aorTo = to.uri;
-            const headers = contentType === 'application/json' && req.body ? JSON.parse(req.body) : {};
             this.logger.info({to, from}, 'startCallRecording request for an outbound call');
 
             const srsUrl = req.get('X-Srs-Url');
@@ -781,49 +873,96 @@ Duration=${payload.duration} `
               headers
             }));
             try {
-              succeeded = (await Promise.all(
-                this.srsClients.map((c) => c.start())
-              )).every((r) => r);
+              this._startRecordingNoAnswerTimer(res);
+              await Promise.any(this.srsClients.map((c) => c.start()));
+              succeeded = true;
             } catch (err) {
               this.logger.error({err}, 'Error starting SipRec call recording');
+              succeeded = false;
             }
           }
           else if (reason === 'stopCallRecording') {
-            if (!this.srsClients.length) {
+            if (!this.srsClients.length || !this.srsClients.some((c) => c.activated)) {
               res.send(400);
               this.logger.info('discarding stopCallRecording request because we are not recording');
               return;
             }
             try {
-              succeeded = (await Promise.all(
-                this.srsClients.map((c) => c.stop())
-              )).every((r) => r);
+              this._startRecordingNoAnswerTimer(res);
+              await Promise.any(this.srsClients.map((c) => {
+                if (c.activated) {
+                  c.stop();
+                }
+              }));
+              succeeded = true;
             } catch (err) {
               this.logger.error({err}, 'Error stopping SipRec call recording');
+              succeeded = false;
             }
             this.srsClients = [];
           }
           else if (reason === 'pauseCallRecording') {
-            if (!this.srsClients.length || this.srsClients.every((c) => c.paused)) {
+            if (!this.srsClients.length || !this.srsClients.some((c) => c.activated && !c.paused)) {
               this.logger.info('discarding invalid pauseCallRecording request');
               res.send(400);
               return;
             }
-            succeeded = (await Promise.all(
-              this.srsClients.map((c) => c.pause())
-            )).every((r) => r);
+            try {
+              this._startRecordingNoAnswerTimer(res);
+              await Promise.any(this.srsClients.map((c) => {
+                if (c.activated && !c.paused) {
+                  c.pause({headers});
+                }
+              }));
+              succeeded = true;
+            } catch (err) {
+              this.logger.error({err}, 'Error pausing SipRec call recording');
+              succeeded = false;
+            }
           }
           else if (reason === 'resumeCallRecording') {
-            if (!this.srsClients.length || !this.srsClients.every((c) => c.paused)) {
+            if (!this.srsClients.length || !this.srsClients.some((c) => c.activated && c.paused)) {
               res.send(400);
               this.logger.info('discarding invalid resumeCallRecording request');
               return;
             }
-            succeeded = (await Promise.all(
-              this.srsClients.map((c) => c.resume())
-            )).every((r) => r);
+            try {
+              this._startRecordingNoAnswerTimer(res);
+              await Promise.any(this.srsClients.map((c) => {
+                if (c.activated && c.paused) {
+                  c.resume({headers});
+                }
+              }));
+              succeeded = true;
+            } catch (err) {
+              this.logger.error({err}, 'Error resuming SipRec call recording');
+              succeeded = false;
+            }
+          }
+          if (!this.isRecordingNoAnswerResponded) {
+            this._clearRecordingNoAnswerTimer();
+            res.send(succeeded ? 200 : 503);
+          }
+        } else if (reason.includes('Dtmf')) {
+          const arr = /Signal=\s*([0-9#*])/.exec(req.body);
+          if (!arr) {
+            this.logger.info({body: req.body}, '_onInfo: invalid INFO Dtmf');
+            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;
+          const dtmfOpts = {
+            ...this.rtpEngineOpts.common,
+            'from-tag': this.rtpEngineOpts.uas.tag,
+            code,
+            duration
+          };
+          const response = await this.playDTMF(dtmfOpts);
+          if ('ok' !== response.result) {
+            this.logger.info({response}, `rtpengine play Dtmf failed with ${JSON.stringify(response)}`);
+            throw new Error('rtpengine failed: answer');
           }
-          res.send(succeeded ? 200 : 503);
         }
       }
       else if (dlg.type === 'uac' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
@@ -861,10 +1000,9 @@ Duration=${payload.duration} `
         }
       }
       else {
-        const immutableHdrs = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
         const headers = {};
         Object.keys(req.headers).forEach((h) => {
-          if (!immutableHdrs.includes(h)) headers[h] = req.headers[h];
+          if (!IMMUTABLE_HEADERS.includes(h)) headers[h] = req.headers[h];
         });
         const response = await dlg.other.request({method: 'INFO', headers, body: req.body});
         const responseHeaders = {};
@@ -880,6 +1018,16 @@ Duration=${payload.duration} `
 
   async _onFeatureServerTransfer(dlg, req, res) {
     try {
+      // the following properties are ignored in the REFER headers
+      // 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,
+        // eslint-disable-next-line no-unused-vars
+        'X-Refer-To-Leave-Untouched': _leave,
+        ...customHeaders
+      } = req.headers;
+
       const referTo = req.getParsedHeader('Refer-To');
       const uri = parseUri(referTo.uri);
       this.logger.info({uri, referTo}, 'received REFER from feature server');
@@ -896,8 +1044,10 @@ Duration=${payload.duration} `
         const response = await this.uac.request({
           method: 'REFER',
           headers: {
-            'Refer-To': stringifyUri(uri),
-            'Referred-By': stringifyUri(u)
+            // Make sure the uri is protected by <> if uri is complex form
+            'Refer-To': `<${stringifyUri(uri)}>`,
+            'Referred-By': `<${stringifyUri(u)}>`,
+            ...customHeaders
           }
         });
         return res.send(response.status);

lib/utils.js

@@ -1,6 +1,11 @@
 const rtpCharacteristics = require('../data/rtp-transcoding');
 const srtpCharacteristics = require('../data/srtp-transcoding');
 const debug = require('debug')('jambonz:sbc-outbound');
+const CIDRMatcher = require('cidr-matcher');
+const dns = require('dns');
+
+const cidrMatcher = process.env.PRIVATE_VOIP_NETWORK_CIDR ?
+  new CIDRMatcher(process.env.PRIVATE_VOIP_NETWORK_CIDR.split(',')) : null;
 
 function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams) {
   const from = req.getParsedHeader('from');
@@ -16,10 +21,14 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams
   const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCopy;
   const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCopy;
 
-  /* webrtc clients (e.g. sipjs) send DMTF via SIP INFO */
-  if ((srcIsUsingSrtp || dstIsUsingSrtp) && !teams) {
-    dstOpts.flags.push('inject DTMF');
-    srcOpts.flags.push('inject DTMF');
+  /** Allow feature server to send DTMF to the call excepts call from/to teams */
+  if (!teams) {
+    if (!dstOpts.flags.includes('inject DTMF')) {
+      dstOpts.flags.push('inject DTMF');
+    }
+    if (!srcOpts.flags.includes('inject DTMF')) {
+      srcOpts.flags.push('inject DTMF');
+    }
   }
   const common = {
     'call-id': req.get('Call-ID'),
@@ -181,6 +190,41 @@ const nudgeCallCounts = async(logger, sids, nudgeOperator, writers) => {
   return {callsSP: null, calls: null, callsApp: null};
 };
 
+const isPrivateVoipNetwork = async(uri) => {
+  if (cidrMatcher) {
+    try {
+      const arr = /sips?:.*@(.*?)(:\d+)?(;.*)$/.exec(uri);
+      if (arr) {
+        const input = arr[1];
+        let addresses;
+        if (input.match(/^\d{1,3}\.\d{1,3}\.\d{1,3}\.\d{1,3}$/)) {
+          addresses = [input];
+        } else {
+          addresses = await dns.resolve4(input);
+        }
+        for (const ip of addresses) {
+          if (cidrMatcher.contains(ip)) {
+            return true;
+          }
+        }
+      }
+    } catch (err) {}
+  }
+  return false;
+};
+
+function makeBlacklistGatewayKey(key) {
+  return `blacklist-sip-gateway:${key}`;
+}
+
+async function isBlackListedSipGateway(client, logger, sip_gateway_sid) {
+  try {
+    return await client.exists(makeBlacklistGatewayKey(sip_gateway_sid)) === 1;
+  } catch (err) {
+    logger.error({err}, `isBlackListedSipGateway: error while checking blacklist for ${sip_gateway_sid}`);
+  }
+}
+
 module.exports = {
   makeRtpEngineOpts,
   selectHostPort,
@@ -190,5 +234,7 @@ module.exports = {
   equalsIgnoreOrder,
   systemHealth,
   createHealthCheckApp,
-  nudgeCallCounts
+  nudgeCallCounts,
+  isPrivateVoipNetwork,
+  isBlackListedSipGateway
 };

package.json

@@ -1,6 +1,6 @@
 {
   "name": "sbc-outbound",
-  "version": "0.8.5",
+  "version": "0.9.0",
   "main": "app.js",
   "engines": {
     "node": ">= 12.0.0"
@@ -24,30 +24,30 @@
     "start": "node app",
     "test": "NODE_ENV=test HTTP_PORT=3050 JAMBONES_HOSTING=1 JAMBONES_NETWORK_CIDR=127.0.0.1/32 JAMBONES_MYSQL_HOST=127.0.0.1 JAMBONES_MYSQL_USER=jambones_test JAMBONES_MYSQL_PASSWORD=jambones_test JAMBONES_MYSQL_DATABASE=jambones_test JAMBONES_REDIS_HOST=localhost JAMBONES_REDIS_PORT=16379 JAMBONES_TIME_SERIES_HOST=127.0.0.1 JAMBONES_LOGLEVEL=error DRACHTIO_SECRET=cymru DRACHTIO_HOST=127.0.0.1 DRACHTIO_PORT=9060 JAMBONES_RTPENGINES=127.0.0.1:12222 node test/ ",
     "coverage": "./node_modules/.bin/nyc --reporter html --report-dir ./coverage npm run test",
-    "jslint": "eslint app.js lib"
+    "jslint": "eslint app.js lib --fix"    
   },
   "dependencies": {
-    "@jambonz/db-helpers": "^0.9.0",
-    "@jambonz/realtimedb-helpers": "^0.8.6",
+    "@jambonz/db-helpers": "^0.9.3",
+    "@jambonz/realtimedb-helpers": "^0.8.8",
     "@jambonz/http-health-check": "^0.0.1",
-    "@jambonz/mw-registrar": "0.2.4",
-    "@jambonz/rtpengine-utils": "^0.4.3",
-    "@jambonz/siprec-client-utils": "^0.2.6",
-    "@jambonz/stats-collector": "^0.1.9",
-    "@jambonz/time-series": "^0.2.5",
+    "@jambonz/mw-registrar": "0.2.7",
+    "@jambonz/rtpengine-utils": "^0.4.4",
+    "@jambonz/siprec-client-utils": "^0.2.7",
+    "@jambonz/stats-collector": "^0.1.10",
+    "@jambonz/time-series": "^0.2.8",
     "cidr-matcher": "^2.1.1",
     "debug": "^4.3.4",
-    "drachtio-fn-b2b-sugar": "^0.0.12",
-    "drachtio-srf": "^4.5.31",
-    "express": "^4.18.1",
-    "pino": "^7.11.0",
-    "sdp-transform": "^2.14.1"
+    "drachtio-fn-b2b-sugar": "^0.1.0",
+    "drachtio-srf": "^4.5.39",
+    "express": "^4.19.2",
+    "pino": "^8.20.0",
+    "sdp-transform": "^2.14.2"
   },
   "devDependencies": {
     "bent": "^7.3.12",
     "eslint": "^7.32.0",
-    "eslint-plugin-promise": "^5.2.0",
+    "eslint-plugin-promise": "^6.1.1",
     "nyc": "^15.1.0",
-    "tape": "^5.5.3"
+    "tape": "^5.7.5"
   }
 }

test/db/jambones-sql.sql

@@ -1,4 +1,5 @@
 /* SQLEditor (MySQL (2))*/
+
 SET FOREIGN_KEY_CHECKS=0;
 
 DROP TABLE IF EXISTS account_static_ips;
@@ -13,6 +14,8 @@ DROP TABLE IF EXISTS beta_invite_codes;
 
 DROP TABLE IF EXISTS call_routes;
 
+DROP TABLE IF EXISTS clients;
+
 DROP TABLE IF EXISTS dns_records;
 
 DROP TABLE IF EXISTS lcr;
@@ -51,6 +54,8 @@ DROP TABLE IF EXISTS signup_history;
 
 DROP TABLE IF EXISTS smpp_addresses;
 
+DROP TABLE IF EXISTS google_custom_voices;
+
 DROP TABLE IF EXISTS speech_credentials;
 
 DROP TABLE IF EXISTS system_information;
@@ -127,6 +132,19 @@ application_sid CHAR(36) NOT NULL,
 PRIMARY KEY (call_route_sid)
 ) COMMENT='a regex-based pattern match for call routing';
 
+CREATE TABLE clients
+(
+client_sid CHAR(36) NOT NULL UNIQUE ,
+account_sid CHAR(36) NOT NULL,
+is_active BOOLEAN NOT NULL DEFAULT 1,
+username VARCHAR(64),
+password VARCHAR(1024),
+allow_direct_app_calling BOOLEAN NOT NULL DEFAULT 1,
+allow_direct_queue_calling BOOLEAN NOT NULL DEFAULT 1,
+allow_direct_user_calling BOOLEAN NOT NULL DEFAULT 1,
+PRIMARY KEY (client_sid)
+);
+
 CREATE TABLE dns_records
 (
 dns_record_sid CHAR(36) NOT NULL UNIQUE ,
@@ -322,9 +340,20 @@ last_tested DATETIME,
 tts_tested_ok BOOLEAN,
 stt_tested_ok BOOLEAN,
 created_at DATETIME NOT NULL DEFAULT CURRENT_TIMESTAMP,
+label VARCHAR(64),
 PRIMARY KEY (speech_credential_sid)
 );
 
+CREATE TABLE google_custom_voices
+(
+google_custom_voice_sid CHAR(36) NOT NULL UNIQUE ,
+speech_credential_sid CHAR(36) NOT NULL,
+model VARCHAR(512) NOT NULL,
+reported_usage ENUM('REPORTED_USAGE_UNSPECIFIED','REALTIME','OFFLINE') DEFAULT 'REALTIME',
+name VARCHAR(64) NOT NULL,
+PRIMARY KEY (google_custom_voice_sid)
+);
+
 CREATE TABLE system_information
 (
 domain_name VARCHAR(255),
@@ -411,7 +440,7 @@ PRIMARY KEY (smpp_gateway_sid)
 CREATE TABLE phone_numbers
 (
 phone_number_sid CHAR(36) UNIQUE ,
-number VARCHAR(132) NOT NULL UNIQUE ,
+number VARCHAR(132) NOT NULL,
 voip_carrier_sid CHAR(36),
 account_sid CHAR(36),
 application_sid CHAR(36),
@@ -424,11 +453,14 @@ CREATE TABLE sip_gateways
 sip_gateway_sid CHAR(36),
 ipv4 VARCHAR(128) NOT NULL COMMENT 'ip address or DNS name of the gateway.  For gateways providing inbound calling service, ip address is required.',
 netmask INTEGER NOT NULL DEFAULT 32,
-port INTEGER NOT NULL DEFAULT 5060 COMMENT 'sip signaling port',
+port INTEGER COMMENT 'sip signaling port',
 inbound BOOLEAN NOT NULL COMMENT 'if true, whitelist this IP to allow inbound calls from the gateway',
 outbound BOOLEAN NOT NULL COMMENT 'if true, include in least-cost routing when placing calls to the PSTN',
 voip_carrier_sid CHAR(36) NOT NULL,
 is_active BOOLEAN NOT NULL DEFAULT 1,
+send_options_ping BOOLEAN NOT NULL DEFAULT 0,
+use_sips_scheme BOOLEAN NOT NULL DEFAULT 0,
+pad_crypto BOOLEAN NOT NULL DEFAULT 0,
 protocol ENUM('udp','tcp','tls', 'tls/srtp') DEFAULT 'udp' COMMENT 'Outbound call protocol',
 PRIMARY KEY (sip_gateway_sid)
 ) COMMENT='A whitelisted sip gateway used for origination/termination';
@@ -465,10 +497,21 @@ messaging_hook_sid CHAR(36) COMMENT 'webhook to call for inbound SMS/MMS ',
 app_json TEXT,
 speech_synthesis_vendor VARCHAR(64) NOT NULL DEFAULT 'google',
 speech_synthesis_language VARCHAR(12) NOT NULL DEFAULT 'en-US',
-speech_synthesis_voice VARCHAR(64),
+speech_synthesis_voice VARCHAR(256),
+speech_synthesis_label VARCHAR(64),
 speech_recognizer_vendor VARCHAR(64) NOT NULL DEFAULT 'google',
 speech_recognizer_language VARCHAR(64) NOT NULL DEFAULT 'en-US',
+speech_recognizer_label VARCHAR(64),
+use_for_fallback_speech BOOLEAN DEFAULT false,
+fallback_speech_synthesis_vendor VARCHAR(64),
+fallback_speech_synthesis_language VARCHAR(12),
+fallback_speech_synthesis_voice VARCHAR(256),
+fallback_speech_synthesis_label VARCHAR(64),
+fallback_speech_recognizer_vendor VARCHAR(64),
+fallback_speech_recognizer_language VARCHAR(64),
+fallback_speech_recognizer_label VARCHAR(64),
 created_at DATETIME NOT NULL DEFAULT CURRENT_TIMESTAMP,
+record_all_calls BOOLEAN NOT NULL DEFAULT false,
 PRIMARY KEY (application_sid)
 ) COMMENT='A defined set of behaviors to be applied to phone calls ';
 
@@ -506,6 +549,9 @@ subspace_client_secret VARCHAR(255),
 subspace_sip_teleport_id VARCHAR(255),
 subspace_sip_teleport_destinations VARCHAR(255),
 siprec_hook_sid CHAR(36),
+record_all_calls BOOLEAN NOT NULL DEFAULT false,
+record_format VARCHAR(16) NOT NULL DEFAULT 'mp3',
+bucket_credential VARCHAR(8192) COMMENT 'credential used to authenticate with storage service',
 PRIMARY KEY (account_sid)
 ) COMMENT='An enterprise that uses the platform for comm services';
 
@@ -526,6 +572,9 @@ ALTER TABLE call_routes ADD FOREIGN KEY account_sid_idxfk_3 (account_sid) REFERE
 
 ALTER TABLE call_routes ADD FOREIGN KEY application_sid_idxfk (application_sid) REFERENCES applications (application_sid);
 
+CREATE INDEX client_sid_idx ON clients (client_sid);
+ALTER TABLE clients ADD CONSTRAINT account_sid_idxfk_13 FOREIGN KEY account_sid_idxfk_13 (account_sid) REFERENCES accounts (account_sid);
+
 CREATE INDEX dns_record_sid_idx ON dns_records (dns_record_sid);
 ALTER TABLE dns_records ADD FOREIGN KEY account_sid_idxfk_4 (account_sid) REFERENCES accounts (account_sid);
 
@@ -590,8 +639,6 @@ CREATE INDEX smpp_address_sid_idx ON smpp_addresses (smpp_address_sid);
 CREATE INDEX service_provider_sid_idx ON smpp_addresses (service_provider_sid);
 ALTER TABLE smpp_addresses ADD FOREIGN KEY service_provider_sid_idxfk_4 (service_provider_sid) REFERENCES service_providers (service_provider_sid);
 
-CREATE UNIQUE INDEX speech_credentials_idx_1 ON speech_credentials (vendor,account_sid);
-
 CREATE INDEX speech_credential_sid_idx ON speech_credentials (speech_credential_sid);
 CREATE INDEX service_provider_sid_idx ON speech_credentials (service_provider_sid);
 ALTER TABLE speech_credentials ADD FOREIGN KEY service_provider_sid_idxfk_5 (service_provider_sid) REFERENCES service_providers (service_provider_sid);
@@ -599,6 +646,10 @@ ALTER TABLE speech_credentials ADD FOREIGN KEY service_provider_sid_idxfk_5 (ser
 CREATE INDEX account_sid_idx ON speech_credentials (account_sid);
 ALTER TABLE speech_credentials ADD FOREIGN KEY account_sid_idxfk_8 (account_sid) REFERENCES accounts (account_sid);
 
+CREATE INDEX google_custom_voice_sid_idx ON google_custom_voices (google_custom_voice_sid);
+CREATE INDEX speech_credential_sid_idx ON google_custom_voices (speech_credential_sid);
+ALTER TABLE google_custom_voices ADD FOREIGN KEY speech_credential_sid_idxfk (speech_credential_sid) REFERENCES speech_credentials (speech_credential_sid) ON DELETE CASCADE;
+
 CREATE INDEX user_sid_idx ON users (user_sid);
 CREATE INDEX email_idx ON users (email);
 CREATE INDEX phone_idx ON users (phone);
@@ -628,6 +679,8 @@ CREATE INDEX smpp_gateway_sid_idx ON smpp_gateways (smpp_gateway_sid);
 CREATE INDEX voip_carrier_sid_idx ON smpp_gateways (voip_carrier_sid);
 ALTER TABLE smpp_gateways ADD FOREIGN KEY voip_carrier_sid_idxfk (voip_carrier_sid) REFERENCES voip_carriers (voip_carrier_sid);
 
+CREATE UNIQUE INDEX phone_numbers_unique_idx_voip_carrier_number ON phone_numbers (number,voip_carrier_sid);
+
 CREATE INDEX phone_number_sid_idx ON phone_numbers (phone_number_sid);
 CREATE INDEX number_idx ON phone_numbers (number);
 CREATE INDEX voip_carrier_sid_idx ON phone_numbers (voip_carrier_sid);

test/scenarios/uac-pcap-carrier-fail-blacklist.xml

@@ -0,0 +1,59 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAC with media">
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:16173333456@127.0.0.1 SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:16173333456@127.0.0.1>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
+      X-Call-Sid: ff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: phone
+      Subject: uac-pcap-carrier-fail-limits
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[local_ip_type] [local_ip]
+      t=0 0
+      m=audio [auto_media_port] RTP/AVP 8 101
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-11,16
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="603" rtd="true" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:sip:+16173333456@127.0.0.1 SIP/2.0
+      [last_Via]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Subject: uac-pcap-carrier-fail-blacklist
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+
+</scenario>

test/scenarios/uac-sip-uri-proxy.xml

@@ -0,0 +1,120 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+<!--                                                                    -->
+<!--                 Sipp 'uac' scenario with pcap (rtp) play           -->
+<!--                                                                    -->
+
+<scenario name="UAC with media">
+  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
+  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:john@foo.bar SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+      To: <sip:john@foo.bar>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: uac-sip-uri-proxy
+      X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
+      X-Call-Sid: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: sip
+      X-SIP-Proxy: 172.39.0.20:5060
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[local_ip_type] [local_ip]
+      t=0 0
+      m=audio [auto_media_port] RTP/AVP 8 101
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-11,16
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+</recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200">
+  </recv>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:john@foo.bar SIP/2.0
+      [last_Via:]
+      [last_From:]
+      To: <sip:sip:john@jambonz.org>[peer_tag_param]
+      [last_Call-ID:]
+      CSeq: 2 ACK
+      Subject: 
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
+  <nop>
+    <action>
+      <exec play_pcap_audio="pcap/g711a.pcap"/>
+    </action>
+  </nop>
+
+  <!-- Pause briefly      -->
+  <pause milliseconds="2000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:john@foo.bar SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 3 BYE
+      Max-Forwards: 70
+      Subject: uac-sip-uri-proxy
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

test/sip-tests.js

@@ -26,6 +26,7 @@ function connect(connectable) {
 test('sbc-outbound tests', async(t) => {
   const {srf} = require('../app');
   const { queryCdrs } = srf.locals;
+  const redisClient = srf.locals.realtimeDbHelpers.client;
 
   try {
     await connect(srf);
@@ -84,6 +85,10 @@ test('sbc-outbound tests', async(t) => {
     await sippUac('uac-sip-uri-auth-success.xml');
     t.pass('successfully connected to sip uri that requires auth');
   
+    /* invite to sipUri through proxy */
+    await sippUac('uac-sip-uri-proxy.xml');
+    t.pass('successfully connected to sip uri through proxy');
+  
     // re-rack test data
     execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/jambones-sql.sql`);
     execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/populate-test-data5.sql`);
@@ -91,12 +96,22 @@ test('sbc-outbound tests', async(t) => {
     /* fails when session limit exceeded */
     await sippUac('uac-pcap-carrier-fail-limits.xml');
     t.pass('fails when max calls in progress');
-      
+
+    // re-rack test data
+    execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/jambones-sql.sql`);
+    execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/populate-test-data.sql`);
+
+    // Black list good carrier for some seconds
+    await redisClient.setex('blacklist-sip-gateway:124a5339-c62c-4075-9e19-f4de70a96597', 3, '');
+    await sippUac('uac-pcap-carrier-fail-blacklist.xml');
+    t.pass('fails when carrier is blacklisted');
+    await redisClient.del('blacklist-sip-gateway:124a5339-c62c-4075-9e19-f4de70a96597');
+
     await waitFor(25);
 
     const res = await queryCdrs({account_sid: 'ed649e33-e771-403a-8c99-1780eabbc803'});
     console.log(`${res.total} cdrs: ${JSON.stringify(res)}`);
-    t.ok(res.total === 8, 'wrote 8 cdrs');
+    t.ok(res.total === 9, 'wrote 9 cdrs');
 
     srf.disconnect();
   } catch (err) {