fix hangup custom headers (#121)

* feat/support inband dtmf

* wip

* wip

* wip

* wip

* wip

* wip

* wip

.github/workflows/docker-publish.yml

@@ -2,6 +2,8 @@ name: Docker
 
 on:
   push:
+    branches:
+      - main
     tags:
       - '*'
 
@@ -18,7 +20,7 @@ jobs:
       - name: prepare tag 
         id: prepare_tag
         run: |
-            IMAGE_ID=$GITHUB_REPOSITORY
+            IMAGE_ID=jambonz/sbc-outbound
 
             # Strip git ref prefix from version
             VERSION=$(echo "${{ github.ref }}" | sed -e 's,.*/\(.*\),\1,')

app.js

@@ -3,7 +3,12 @@ assert.ok(process.env.JAMBONES_MYSQL_HOST &&
   process.env.JAMBONES_MYSQL_USER &&
   process.env.JAMBONES_MYSQL_PASSWORD &&
   process.env.JAMBONES_MYSQL_DATABASE, 'missing JAMBONES_MYSQL_XXX env vars');
-assert.ok(process.env.JAMBONES_REDIS_HOST, 'missing JAMBONES_REDIS_HOST env var');
+if (process.env.JAMBONES_REDIS_SENTINELS) {
+  assert.ok(process.env.JAMBONES_REDIS_SENTINEL_MASTER_NAME,
+    'missing JAMBONES_REDIS_SENTINEL_MASTER_NAME env var, JAMBONES_REDIS_SENTINEL_PASSWORD env var is optional');
+} else {
+  assert.ok(process.env.JAMBONES_REDIS_HOST, 'missing JAMBONES_REDIS_HOST env var');
+}
 assert.ok(process.env.DRACHTIO_PORT || process.env.DRACHTIO_HOST, 'missing DRACHTIO_PORT env var');
 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');
@@ -61,11 +66,8 @@ const {
   incrKey,
   decrKey,
   retrieveSet,
-  isMemberOfSet
-} = require('@jambonz/realtimedb-helpers')({
-  host: process.env.JAMBONES_REDIS_HOST,
-  port: process.env.JAMBONES_REDIS_PORT || 6379
-}, logger);
+  isMemberOfSet,
+} = require('@jambonz/realtimedb-helpers')({}, logger);
 
 const activeCallIds = new Map();
 const Emitter = require('events');
@@ -102,10 +104,7 @@ srf.locals = {...srf.locals,
     isMemberOfSet
   }
 };
-const {initLocals, checkLimits, route} = require('./lib/middleware')(srf, logger, {
-  host: process.env.JAMBONES_REDIS_HOST,
-  port: process.env.JAMBONES_REDIS_PORT || 6379
-});
+const {initLocals, checkLimits, route} = require('./lib/middleware')(srf, logger, redisClient);
 const ngProtocol = process.env.JAMBONES_NG_PROTOCOL || 'udp';
 const ngPort = process.env.RTPENGINE_PORT || ('udp' === ngProtocol ? 22222 : 8080);
 const {getRtpEngine, setRtpEngines} = require('@jambonz/rtpengine-utils')([], logger, {

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} = 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}`;
 };
@@ -33,19 +39,21 @@ const createBLegToHeader = (req, teams) => {
   return `sip:anonymous@${host}`;
 };
 
-const initCdr = (srf, req) => {
-  const uri = parseUri(req.uri);
+const initCdr = (req, invite) => {
+  const {srf} = req;
+  const {trace_id} = req.locals;
+  const uri = parseUri(invite.uri);
   const regex = /^\+(\d+)$/;
-  let arr = regex.exec(req.calledNumber);
-  const to = arr ? arr[1] : req.calledNumber;
-  arr = regex.exec(req.callingNumber);
-  const from = arr ? arr[1] : req.callingNumber;
+  let arr = regex.exec(invite.calledNumber);
+  const to = arr ? arr[1] : invite.calledNumber;
+  arr = regex.exec(invite.callingNumber);
+  const from = arr ? arr[1] : invite.callingNumber;
   const applicationSid = req.get('X-Application-Sid');
 
   return {
     account_sid: req.get('X-Account-Sid'),
     call_sid: req.get('X-Call-Sid'),
-    sip_callid: req.get('Call-ID'),
+    sip_callid: invite.get('Call-ID'),
     ...(applicationSid && {application_sid: applicationSid}),
     from,
     to,
@@ -55,7 +63,7 @@ const initCdr = (srf, req) => {
     direction: 'outbound',
     host: srf.locals.sipAddress,
     remote_host: uri.host,
-    trace_id: req.get('X-Trace-ID') || '00000000000000000000000000000000'
+    trace_id: trace_id || '00000000000000000000000000000000'
   };
 };
 
@@ -97,6 +105,10 @@ class CallSession extends Emitter {
     this._mediaReleased = false;
   }
 
+  get service_provider_sid() {
+    return this.req.locals.service_provider_sid;
+  }
+
   get account_sid() {
     return this.req.locals.account_sid;
   }
@@ -169,8 +181,9 @@ class CallSession extends Emitter {
     this.subscribeRequest = subscribeRequest;
     this.subscribeAnswer = subscribeAnswer;
     this.unsubscribe = unsubscribe;
+    this.srsClients = [];
 
-    this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, teams);
+    this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, false, teams);
     this.rtpEngineResource = {destroy: this.del.bind(null, this.rtpEngineOpts.common)};
     let proxy, uris;
     const mapGateways = new Map();
@@ -197,21 +210,37 @@ 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`
@@ -227,7 +256,19 @@ class CallSession extends Emitter {
             voip_carrier_sid = await this.lookupCarrierByAccountLcr(account_sid, this.req.calledNumber);
           }
           if (!voip_carrier_sid) {
-            /* no LCR for this account/SP - at this point its a random shuffle of outbound carriers */
+            /* no LCR for this account/SP - try with inbound carrier */
+            const inbound_carrier_sid = this.req.get('X-Voip-Carrier-Sid');
+            if (inbound_carrier_sid) {
+              const gateways = await this.lookupSipGatewaysByCarrier(inbound_carrier_sid);
+              const gws = (gateways || [])
+                .filter((gw) => gw.outbound);
+              if (gws.length) {
+                voip_carrier_sid = inbound_carrier_sid;
+              }
+            }
+          }
+          if (!voip_carrier_sid) {
+            /* no LCR/ inbound carrier for this account/SP - at this point its a random shuffle of outbound carriers */
             voip_carrier_sid = await this.lookupOutboundCarrierForAccount(this.account_sid);
           }
           if (!voip_carrier_sid) {
@@ -257,7 +298,8 @@ class CallSession extends Emitter {
                 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 = {
@@ -274,10 +316,17 @@ class CallSession extends Emitter {
                  * but it will require more significant changes and right now it seems
                  * like a rare use case -- encryption is usually an all or nothing requirement.
                  */
-                this.logger.info({u}, 'using SRTP for outbound call');
-                this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, true);
+                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);
               }
             });
+            // 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 {
@@ -292,15 +341,22 @@ 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) {
@@ -315,7 +371,17 @@ 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) {
@@ -333,9 +399,10 @@ class CallSession extends Emitter {
             this.logger.info({err}, 'Error retrieving iip key');
           }
         }
+        // INVITE request line and To header should be the same.
+        hdrs = {...hdrs, 'To': uri};
         if (gw) {
           this.logger.info({gw}, `sending INVITE to ${uri} via carrier ${gw.name}`);
-          hdrs = {...hdrs, 'To': uri};
           if (gw.diversion) {
             let div = gw.diversion;
             if (div.startsWith('+')) {
@@ -347,6 +414,12 @@ 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 {
@@ -366,8 +439,10 @@ class CallSession extends Emitter {
               '-X-Jambonz-Routing',
               '-X-Jambonz-FS-UUID',
               '-X-Preferred-From-User',
-              'X-Preferred-From-Host',
+              '-X-Preferred-From-Host',
               '-X-Jambonz-FS-UUID',
+              '-X-Voip-Carrier-Sid',
+              '-X-SIP-Proxy'
             ],
             proxyResponseHeaders: [
               'all',
@@ -385,7 +460,7 @@ 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);
@@ -411,7 +486,7 @@ class CallSession extends Emitter {
               }
               else {
                 this.req.locals.cdr = {
-                  ...initCdr(this.req.srf, inv),
+                  ...initCdr(this.req, inv),
                   service_provider_sid: this.req.locals.service_provider_sid,
                   account_sid: this.req.locals.account_sid,
                   ...(this.req.locals.application_sid && {application_sid: this.req.locals.application_sid}),
@@ -500,13 +575,18 @@ 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) {}
 
@@ -527,12 +607,18 @@ class CallSession extends Emitter {
         /* write cdr for connected call */
         if (this.req.locals.cdr) {
           const now = Date.now();
+          const day = new Date();
+          const recordAllCalls = this.req.locals.record_all_calls;
+          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}`;
           this.writeCdrs({...this.req.locals.cdr,
             terminated_at: now,
             termination_reason: dlg.type === 'uas' ? 'caller hungup' : 'called party hungup',
             sip_status: 200,
             answered: true,
-            duration: Math.floor((now - callStart) / 1000)
+            duration: Math.floor((now - callStart) / 1000),
+            ...(recordAllCalls && {recording_url})
           }).catch((err) => this.logger.error({err}, 'Error writing cdr for completed call'));
         }
         /* de-link the 2 Dialogs for GC */
@@ -541,10 +627,7 @@ class CallSession extends Emitter {
         dlg.other = null;
         other.other = null;
 
-        if (this.srsClient) {
-          this.srsClient.stop();
-          this.srsClient = null;
-        }
+        this._stopRecording();
 
         this.logger.info(`call ended with normal termination, there are ${this.activeCallIds.size} active`);
         if (this.activeCallIds.size === 0) this.idleEmitter.emit('idle');
@@ -569,6 +652,13 @@ class CallSession extends Emitter {
     forwardInDialogRequests(uac,  ['notify', 'options', 'message']);
   }
 
+  _stopRecording() {
+    if (this.srsClients.length) {
+      this.srsClients.forEach((c) => c.stop());
+      this.srsClients = [];
+    }
+  }
+
   async _onRefer(dlg, req, res) {
     /* REFER coming in from a sip device, forward to feature server */
     try {
@@ -707,6 +797,7 @@ Duration=${payload.duration} `
             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');
@@ -714,7 +805,7 @@ Duration=${payload.duration} `
             const callSid = req.get('X-Call-Sid');
             const accountSid = req.get('X-Account-Sid');
             const applicationSid = req.get('X-Application-Sid');
-            if (this.srsClient) {
+            if (this.srsClients.length) {
               res.send(400);
               this.logger.info('discarding duplicate startCallRecording request for a call');
               return;
@@ -724,13 +815,14 @@ Duration=${payload.duration} `
               res.send(400);
               return;
             }
-            this.srsClient = new SrsClient(this.logger, {
+            const arr = srsUrl.split(',');
+            this.srsClients = arr.map((url) => new SrsClient(this.logger, {
               srf: dlg.srf,
               direction: 'outbound',
               originalInvite: this.req,
               callingNumber: this.req.callingNumber,
               calledNumber: this.req.calledNumber,
-              srsUrl,
+              srsUrl: url,
               srsRecordingId,
               callSid,
               accountSid,
@@ -744,44 +836,73 @@ Duration=${payload.duration} `
               del: this.del,
               blockMedia: this.blockMedia,
               unblockMedia: this.unblockMedia,
-              unsubscribe: this.unsubscribe
-            });
+              unsubscribe: this.unsubscribe,
+              headers
+            }));
             try {
-              succeeded = await this.srsClient.start();
+              succeeded = (await Promise.all(
+                this.srsClients.map((c) => c.start())
+              )).every((r) => r);
             } catch (err) {
               this.logger.error({err}, 'Error starting SipRec call recording');
             }
           }
           else if (reason === 'stopCallRecording') {
-            if (!this.srsClient) {
+            if (!this.srsClients.length) {
               res.send(400);
               this.logger.info('discarding stopCallRecording request because we are not recording');
               return;
             }
             try {
-              succeeded = await this.srsClient.stop();
+              succeeded = (await Promise.all(
+                this.srsClients.map((c) => c.stop())
+              )).every((r) => r);
             } catch (err) {
               this.logger.error({err}, 'Error stopping SipRec call recording');
             }
-            this.srsClient = null;
+            this.srsClients = [];
           }
           else if (reason === 'pauseCallRecording') {
-            if (!this.srsClient || this.srsClient.paused) {
+            if (!this.srsClients.length || this.srsClients.every((c) => c.paused)) {
               this.logger.info('discarding invalid pauseCallRecording request');
               res.send(400);
               return;
             }
-            succeeded = await this.srsClient.pause();
+            succeeded = (await Promise.all(
+              this.srsClients.map((c) => c.pause())
+            )).every((r) => r);
           }
           else if (reason === 'resumeCallRecording') {
-            if (!this.srsClient || !this.srsClient.paused) {
+            if (!this.srsClients.length || !this.srsClients.every((c) => c.paused)) {
               res.send(400);
               this.logger.info('discarding invalid resumeCallRecording request');
               return;
             }
-            succeeded = await this.srsClient.resume();
+            succeeded = (await Promise.all(
+              this.srsClients.map((c) => c.resume())
+            )).every((r) => r);
           }
           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 ${JSON.stringify(response)}`);
+            throw new Error('rtpengine failed: answer');
+          }
         }
       }
       else if (dlg.type === 'uac' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
@@ -819,15 +940,16 @@ Duration=${payload.duration} `
         }
       }
       else {
-        const response = await dlg.other.request({
-          method: 'INFO',
-          headers: req.headers,
-          body: req.body
-        });
-        res.send(response.status, {
-          headers: response.headers,
-          body: response.body
+        const headers = {};
+        Object.keys(req.headers).forEach((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 = {};
+        if (response.has('Content-Type')) {
+          Object.assign(responseHeaders, {'Content-Type': response.get('Content-Type')});
+        }
+        res.send(response.status, {headers: responseHeaders, body: response.body});
       }
     } catch (err) {
       this.logger.info({err}, `Error handing INFO request on ${dlg.type} leg`);
@@ -865,35 +987,38 @@ Duration=${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 dlg = await this.srf.createUAC(referTo.uri, {localSdp: dlg.local.sdp, headers});
-      this.uas = dlg;
-      this.uas.other = this.uac;
-      this.uac.other = this.uas;
-      this.uas.on('modify', this._onReinvite.bind(this, this.uas));
-      this.uas.on('refer', this._onFeatureServerTransfer.bind(this, this.uas));
-      this.uas.on('destroy', () => {
+
+      const uac = await this.srf.createUAC(referTo.uri, {localSdp: dlg.local.sdp, headers});
+      this.uas = uac;
+      uac.type = 'uas';
+      uac.other = this.uac;
+      this.uac.other = uac;
+      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', () => {
         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');
-        this.uas.other.destroy();
+        uac.other.destroy();
         this.srf.endSession(this.req);
       });
 
-      // modify rtpengine to stream to new feature server
-      let response = await this.offer(Object.assign(this.rtpEngineOpts.offer, {sdp: this.uas.remote.sdp}));
+      const opts = {
+        ...this.rtpEngineOpts.common,
+        'from-tag': this.rtpEngineOpts.uas.tag,
+        sdp: uac.remote.sdp,
+        flags: ['port latching']
+      };
+      const response = await this.offer(opts);
       if ('ok' !== response.result) {
-        throw new Error(`_onReinvite: rtpengine failed: offer: ${JSON.stringify(response)}`);
-      }
-      const sdp = await this.uas.other.modify(response.sdp);
-      const opts = Object.assign({sdp, 'to-tag': res.getParsedHeader('To').params.tag},
-        this.rtpEngineOpts.answer);
-      response = await this.answer(opts);
-      if ('ok' !== response.result) {
-        throw new Error(`_onReinvite: rtpengine failed: ${JSON.stringify(response)}`);
+        throw new Error(`_onFeatureServerTransfer: rtpengine offer failed: ${JSON.stringify(response)}`);
       }
+      dlg.destroy().catch(() => {});
       this.logger.info('successfully moved call to new feature server');
     } catch (err) {
+      res.send(488);
       this.logger.error(err, 'Error handling refer from feature server');
     }
   }

lib/middleware.js

@@ -4,10 +4,10 @@ const Registrar = require('@jambonz/mw-registrar');
 const {selectHostPort, nudgeCallCounts} = require('./utils');
 const FS_UUID_SET_NAME =  'fsUUIDs';
 
-module.exports = (srf, logger, opts) => {
+module.exports = (srf, logger, redisClient) => {
   const {incrKey, decrKey, isMemberOfSet} = srf.locals.realtimeDbHelpers;
   const {stats} = srf.locals;
-  const registrar = new Registrar(opts);
+  const registrar = new Registrar(logger, redisClient);
   const {
     lookupAccountCapacitiesBySid,
     lookupAccountBySid,
@@ -19,6 +19,7 @@ module.exports = (srf, logger, opts) => {
     const callId = req.get('Call-ID');
     req.locals.account_sid = req.get('X-Account-Sid');
     req.locals.application_sid = req.get('X-Application-Sid');
+    req.locals.record_all_calls = req.get('X-Record-All-Calls');
     const traceId = req.locals.trace_id = req.get('X-Trace-ID');
     req.locals.logger = logger.child({
       callId,

lib/utils.js

@@ -1,19 +1,34 @@
 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');
 
-function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
+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');
   const rtpCopy = JSON.parse(JSON.stringify(rtpCharacteristics));
   const srtpCopy = JSON.parse(JSON.stringify(srtpCharacteristics));
+
+  if (padCrypto) {
+    srtpCopy['default'].flags.push('SDES-pad');
+    srtpCopy['teams'].flags.push('SDES-pad');
+  }
+
   const srtpOpts = teams ? srtpCopy['teams'] : srtpCopy['default'];
   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'),
@@ -175,6 +190,29 @@ 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;
+};
+
 module.exports = {
   makeRtpEngineOpts,
   selectHostPort,
@@ -184,5 +222,6 @@ module.exports = {
   equalsIgnoreOrder,
   systemHealth,
   createHealthCheckApp,
-  nudgeCallCounts
+  nudgeCallCounts,
+  isPrivateVoipNetwork
 };

package.json

@@ -1,6 +1,6 @@
 {
   "name": "sbc-outbound",
-  "version": "0.8.3",
+  "version": "0.8.5",
   "main": "app.js",
   "engines": {
     "node": ">= 12.0.0"
@@ -27,18 +27,18 @@
     "jslint": "eslint app.js lib"
   },
   "dependencies": {
-    "@jambonz/db-helpers": "^0.8.0",
-    "@jambonz/realtimedb-helpers": "^0.7.2",
+    "@jambonz/db-helpers": "^0.9.0",
+    "@jambonz/realtimedb-helpers": "^0.8.7",
     "@jambonz/http-health-check": "^0.0.1",
-    "@jambonz/mw-registrar": "0.2.2",
+    "@jambonz/mw-registrar": "0.2.4",
     "@jambonz/rtpengine-utils": "^0.4.3",
-    "@jambonz/siprec-client-utils": "^0.2.4",
-    "@jambonz/stats-collector": "^0.1.8",
+    "@jambonz/siprec-client-utils": "^0.2.6",
+    "@jambonz/stats-collector": "^0.1.9",
     "@jambonz/time-series": "^0.2.5",
     "cidr-matcher": "^2.1.1",
     "debug": "^4.3.4",
     "drachtio-fn-b2b-sugar": "^0.0.12",
-    "drachtio-srf": "^4.5.21",
+    "drachtio-srf": "^4.5.31",
     "express": "^4.18.1",
     "pino": "^7.11.0",
     "sdp-transform": "^2.14.1"

test/scenarios/uac-pcap-inbound-carrier-success.xml

@@ -0,0 +1,92 @@
+<?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: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: phone
+      X-Voip-Carrier-Sid: 287c1452-620d-4195-9f19-c9814ef90d78
+      Subject: uac-pcap-carrier-success
+      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" rtd="true" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip: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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Subject: uac-pcap-carrier-success
+      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: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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Max-Forwards: 70
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+</scenario>

test/scenarios/uac-pcap-sip-routing-success.xml

@@ -0,0 +1,91 @@
+<?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@172.39.0.20 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@172.39.0.20>
+      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: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: sip
+      Subject: uac-pcap-carrier-success
+      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" rtd="true" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip: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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Subject: uac-pcap-carrier-success
+      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: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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Max-Forwards: 70
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+</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

@@ -44,6 +44,14 @@ test('sbc-outbound tests', async(t) => {
     await sippUac('uac-pcap-carrier-success.xml');
     t.pass('successfully completed outbound call to sip trunk');
 
+    /* call to Sip URI with no lcr configured */
+    await sippUac('uac-pcap-sip-routing-success.xml');
+    t.pass('successfully completed outbound call to sip routing trunk');
+
+    /* call to PSTN with no lcr configured */
+    await sippUac('uac-pcap-inbound-carrier-success.xml');
+    t.pass('successfully completed outbound call to sip trunk');
+
     /* call to PSTN with request uri we see in kubernetes */
     await sippUac('uac-pcap-carrier-success-k8s.xml');
     t.pass('successfully completed outbound call to sip trunk (k8S req uri)');
@@ -76,6 +84,10 @@ 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`);
@@ -88,7 +100,7 @@ test('sbc-outbound tests', async(t) => {
 
     const res = await queryCdrs({account_sid: 'ed649e33-e771-403a-8c99-1780eabbc803'});
     console.log(`${res.total} cdrs: ${JSON.stringify(res)}`);
-    t.ok(res.total === 6, 'wrote 6 cdrs');
+    t.ok(res.total === 9, 'wrote 9 cdrs');
 
     srf.disconnect();
   } catch (err) {