bump version to 0.7.6

* initial changes for handling SIP INFO from webrtc clients

* SIP INFO DTMF handling: if media has been released, just relay SIP INFO to FS otherwise transcode to RFC 2833

Dockerfile

@@ -1,10 +1,23 @@
-FROM node: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/
+
+FROM base as build
+
 COPY package.json package-lock.json ./
+
 RUN npm ci
-RUN npm prune
-COPY . /opt/app
+
+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" ]

app.js

@@ -111,6 +111,7 @@ const activeCallIds = srf.locals.activeCallIds;
 
 const {
   initLocals,
+  handleSipRec,
   identifyAccount,
   checkLimits,
   challengeDeviceCalls
@@ -161,7 +162,13 @@ if (process.env.NODE_ENV === 'test') {
 }
 
 /* install middleware */
-srf.use('invite', [initLocals, identifyAccount, checkLimits, challengeDeviceCalls]);
+srf.use('invite', [
+  initLocals,
+  handleSipRec,
+  identifyAccount,
+  checkLimits,
+  challengeDeviceCalls
+]);
 
 srf.invite((req, res) => {
   if (req.has('Replaces')) {

lib/call-session.js

@@ -1,4 +1,5 @@
 const Emitter = require('events');
+const SrsClient = require('@jambonz/siprec-client-utils');
 const {makeRtpEngineOpts, SdpWantsSrtp, makeCallCountKey} = require('./utils');
 const {forwardInDialogRequests} = require('drachtio-fn-b2b-sugar');
 const {parseUri, stringifyUri, SipError} = require('drachtio-srf');
@@ -17,6 +18,20 @@ const createBLegFromHeader = (req) => {
   return '<sip:anonymous@localhost>';
 };
 
+const createSiprecBody = (headers, sdp, type, content) => {
+  const sep = 'uniqueBoundary';
+  headers['Content-Type'] = `multipart/mixed;boundary="${sep}"`;
+  return `--${sep}\r
+Content-Type: application/sdp\r
+\r
+${sdp}\r
+--${sep}\r
+Content-Type: ${type}\r
+Content-Disposition: recording-session\r
+\r
+${content}`;
+};
+
 class CallSession extends Emitter {
   constructor(logger, req, res) {
     super();
@@ -24,6 +39,8 @@ class CallSession extends Emitter {
     this.res = res;
     this.srf = req.srf;
     this.logger = logger.child({callId: req.get('Call-ID')});
+    this.siprec = req.locals.siprec;
+    this.xml = req.locals.xml;
 
     this.getRtpEngine = req.srf.locals.getRtpEngine;
     this.getFeatureServer = req.srf.locals.getFeatureServer;
@@ -33,6 +50,7 @@ class CallSession extends Emitter {
 
     this.decrKey = req.srf.locals.realtimeDbHelpers.decrKey;
     this.callCountKey = makeCallCountKey(req.locals.account_sid);
+    this._mediaReleased = false;
   }
 
   get isFromMSTeams() {
@@ -43,7 +61,17 @@ class CallSession extends Emitter {
     return this.srf.locals.privateSipAddress;
   }
 
+  get isMediaReleased() {
+    return this._mediaReleased;
+  }
+
+  get callerIsUsingSrtp() {
+    const tp = this.rtpEngineOpts?.uas?.mediaOpts['transport-protocol'];
+    return tp && -1 !== tp.indexOf('SAVP');
+  }
+
   async connect() {
+    const {sdp} = this.req.locals;
     this.logger.info('inbound call accepted for routing');
     const engine = this.getRtpEngine();
     if (!engine) {
@@ -61,8 +89,12 @@ class CallSession extends Emitter {
       unblockMedia,
       blockDTMF,
       unblockDTMF,
+      playDTMF,
       subscribeDTMF,
-      unsubscribeDTMF
+      unsubscribeDTMF,
+      subscribeRequest,
+      subscribeAnswer,
+      unsubscribe
     } = engine;
     this.offer = offer;
     this.answer = answer;
@@ -71,8 +103,12 @@ 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;
 
     const featureServer = await this.getFeatureServer();
     if (!featureServer) {
@@ -83,7 +119,7 @@ class CallSession extends Emitter {
     }
     this.logger.debug(`using feature server ${featureServer}`);
 
-    this.rtpEngineOpts = makeRtpEngineOpts(this.req, SdpWantsSrtp(this.req.body), false, this.isFromMSTeams);
+    this.rtpEngineOpts = makeRtpEngineOpts(this.req, SdpWantsSrtp(sdp), false, this.isFromMSTeams);
     this.rtpEngineResource = {destroy: this.del.bind(null, this.rtpEngineOpts.common)};
     const obj = parseUri(this.req.uri);
     let proxy, host, uri;
@@ -108,7 +144,7 @@ class CallSession extends Emitter {
         ...this.rtpEngineOpts.uac.mediaOpts,
         'from-tag': this.rtpEngineOpts.uas.tag,
         direction:  ['public', 'private'],
-        sdp: this.req.body
+        sdp
       };
       const response = await this.offer(opts);
       this.logger.debug({opts, response}, 'response from rtpengine to offer');
@@ -117,7 +153,6 @@ class CallSession extends Emitter {
         throw new Error('rtpengine failed: answer');
       }
 
-      // now send the INVITE in towards the feature servers
       let headers = {
         'From': createBLegFromHeader(this.req),
         'To': this.req.get('To'),
@@ -127,6 +162,10 @@ class CallSession extends Emitter {
       };
       if (this.privateSipAddress) headers = {...headers, Contact: `<sip:${this.privateSipAddress}>`};
 
+      const spdOfferB = this.siprec && this.xml ?
+        createSiprecBody(headers, response.sdp, this.xml.type, this.xml.content) :
+        response.sdp;
+
       const responseHeaders = {};
       if (this.req.locals.carrier) {
         Object.assign(headers, {
@@ -158,6 +197,7 @@ class CallSession extends Emitter {
 
       if (this.req.canceled) throw new Error('call canceled');
 
+      // now send the INVITE in towards the feature servers
       debug(`sending INVITE to ${proxy} with ${uri}`);
       const {uas, uac} = await this.srf.createB2BUA(this.req, this.res, uri, {
         proxy,
@@ -172,7 +212,7 @@ class CallSession extends Emitter {
           '-X-Subspace-Forwarded-For'
         ],
         proxyResponseHeaders: ['all', '-X-Trace-ID'],
-        localSdpB: response.sdp,
+        localSdpB: spdOfferB,
         localSdpA: async(sdp, res) => {
           this.rtpEngineOpts.uac.tag = res.getParsedHeader('To').params.tag;
           const opts = {
@@ -224,6 +264,12 @@ class CallSession extends Emitter {
       this.rtpEngineResource.destroy().catch((err) => {});
       this.activeCallIds.delete(callId);
       if (dlg.other && dlg.other.connected) dlg.other.destroy().catch((e) => {});
+
+      if (this.srsClient) {
+        this.srsClient.stop();
+        this.srsClient = null;
+      }
+
       this.srf.endSession(this.req);
     });
 
@@ -288,6 +334,11 @@ class CallSession extends Emitter {
         dlg.other = null;
         other.other = null;
 
+        if (this.srsClient) {
+          this.srsClient.stop();
+          this.srsClient = null;
+        }
+
         this.logger.info(`call ended with normal termination, there are ${this.activeCallIds.size} active`);
         this.srf.endSession(this.req);
       });
@@ -344,7 +395,19 @@ Duration=${payload.duration} `
    */
   async replaces(req, res) {
     try {
-      let opts = Object.assign({}, this.rtpEngineOpts.uas.mediaOpts, {sdp: req.body});
+      const fromTag = this.rtpEngineOpts.uas.tag;
+      const toTag =  this.rtpEngineOpts.uac.tag;
+      const offerMedia = this.rtpEngineOpts.uac.mediaOpts;
+      const answerMedia = this.rtpEngineOpts.uas.mediaOpts;
+      const direction = ['public', 'private'];
+      let opts = {
+        ...this.rtpEngineOpts.common,
+        ...offerMedia,
+        'from-tag': fromTag,
+        'to-tag': toTag,
+        direction,
+        sdp: req.body,
+      };
       let response = await this.offer(opts);
       if ('ok' !== response.result) {
         res.send(488);
@@ -352,8 +415,13 @@ Duration=${payload.duration} `
       }
       this.logger.info({opts, response}, 'sent offer for reinvite to rtpengine');
       const sdp = await this.uac.modify(response.sdp);
-      opts = Object.assign({}, this.rtpEngineOpts.uac.mediaOpts, {sdp, 'to-tag': this.toTag});
-      Object.assign(this.rtpEngineOpts.uas.mediaOpts, {'to-tag': this.toTag});
+      opts = {
+        ...this.rtpEngineOpts.common,
+        ...answerMedia,
+        'from-tag': fromTag,
+        'to-tag': toTag,
+        sdp
+      };
       response = await this.answer(opts);
       if ('ok' !== response.result) {
         res.send(488);
@@ -399,6 +467,10 @@ Duration=${payload.duration} `
         res.send(200, {body: dlg.local.sdp});
         return;
       }
+      const offeredSdp = Array.isArray(req.payload) && req.payload.length > 1 ?
+        req.payload.find((p) => p.type === 'application/sdp').content :
+        req.body;
+
       const reason = req.get('X-Reason');
       const fromTag = dlg.type === 'uas' ? this.rtpEngineOpts.uas.tag : this.rtpEngineOpts.uac.tag;
       const toTag = dlg.type === 'uas' ? this.rtpEngineOpts.uac.tag : this.rtpEngineOpts.uas.tag;
@@ -411,9 +483,9 @@ Duration=${payload.duration} `
         'from-tag': fromTag,
         'to-tag': toTag,
         direction,
-        sdp: req.body,
+        sdp: offeredSdp,
       };
-      //if (reason && opts.flags && !opts.flags.includes('reset')) opts.flags.push('reset');
+      if (reason && opts.flags && !opts.flags.includes('reset')) opts.flags.push('reset');
 
       let response = await this.offer(opts);
       if ('ok' !== response.result) {
@@ -423,9 +495,12 @@ Duration=${payload.duration} `
 
       /* if this is a re-invite from the FS to change media anchoring, avoid sending the reinvite out */
       let sdp;
-      if (reason && dlg.type === 'uac' && ['release-media', 'anchor-media'].includes(reason)) {
+      if (reason && dlg.type === 'uac' && ['release-media', 'anchor-media'].includes(reason) &&
+        !this.callerIsUsingSrtp) {
         this.logger.info({response}, `got a reinvite from FS to ${reason}`);
         sdp = dlg.other.remote.sdp;
+        answerMedia.flags = ['asymmetric', 'port latching'];
+        this._mediaReleased = 'release-media' === reason;
       }
       else {
         sdp = await dlg.other.modify(response.sdp);
@@ -450,6 +525,8 @@ Duration=${payload.duration} `
 
   async _onInfo(dlg, req, res) {
     const fromTag = dlg.type === 'uas' ? this.rtpEngineOpts.uas.tag : this.rtpEngineOpts.uac.tag;
+    const toTag = dlg.type === 'uas' ? this.rtpEngineOpts.uac.tag : this.rtpEngineOpts.uas.tag;
+    const contentType = req.get('Content-Type');
     try {
       if (dlg.type === 'uac' && req.has('X-Reason')) {
         const reason = req.get('X-Reason');
@@ -459,16 +536,135 @@ Duration=${payload.duration} `
           'from-tag': fromTag
         };
         this.logger.info(`_onInfo: got request ${reason}`);
-        res.send(200);
 
         if (reason.startsWith('mute')) {
           const response = Promise.all([this.blockMedia(opts), this.blockDTMF(opts)]);
+          res.send(200);
           this.logger.info({response}, `_onInfo: response to rtpengine command for ${reason}`);
         }
         else if (reason.startsWith('unmute')) {
           const response = Promise.all([this.unblockMedia(opts), this.unblockDTMF(opts)]);
+          res.send(200);
           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 a 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: 'inbound',
+              originalInvite: this.req,
+              callingNumber: this.req.callingNumber,
+              calledNumber: this.req.calledNumber,
+              srsUrl,
+              srsRecordingId,
+              callSid,
+              accountSid,
+              applicationSid,
+              rtpEngineOpts: this.rtpEngineOpts,
+              fromTag,
+              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 === 'uas' && ['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.uas.tag,
+            code,
+            duration
+          };
+          const response = await this.playDTMF(opts);
+          if ('ok' !== response.result) {
+            this.logger.info({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
+            throw new Error('rtpengine failed: answer');
+          }
+          res.send(200);
+        }
       }
       else {
         const immutableHdrs = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
@@ -484,6 +680,10 @@ Duration=${payload.duration} `
         res.send(response.status, {headers: responseHeaders, body: response.body});
       }
     } catch (err) {
+      if (this.srsClient) {
+        this.srsClient = null;
+      }
+      res.send(500);
       this.logger.info({err}, `Error handing INFO request on ${dlg.type} leg`);
     }
   }
@@ -520,11 +720,19 @@ Duration=${payload.duration} `
         if (e164 && !uri.user.startsWith('+')) {
           uri.user = `+${uri.user}`;
         }
+        // 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,
+          ...customHeaders
+        } = req.headers;
+
         const response = await this.uas.request({
           method: 'REFER',
           headers: {
             'Refer-To': stringifyUri(uri),
-            'Referred-By': stringifyUri(u)
+            'Referred-By': stringifyUri(u),
+            ...customHeaders
           }
         });
         return res.send(response.status);

lib/middleware.js

@@ -70,7 +70,8 @@ module.exports = function(srf, logger) {
   });
 
   const initLocals = (req, res, next) => {
-    req.locals = req.locals || {};
+    const callId = req.get('Call-ID');
+    req.locals = req.locals || {callId};
 
     /* check if forwarded by a proxy that applied an X-Forwarded-For Header */
     if (req.has('X-Forwarded-For') || req.has('X-Subspace-Forwarded-For')) {
@@ -83,7 +84,6 @@ module.exports = function(srf, logger) {
       req.source_address = original_source_address;
     }
     req.locals.cdr = initCdr(req);
-    const callId = req.get('Call-ID');
     req.on('cancel', () => {
       logger.info({callId}, 'caller hungup before connecting to feature server');
       req.canceled = true;
@@ -109,8 +109,31 @@ module.exports = function(srf, logger) {
     next();
   };
 
+  const handleSipRec = async(req, res, next) => {
+    const {callId} = req.locals;
+    if (Array.isArray(req.payload) && req.payload.length > 1) {
+      const sdp = req.payload
+        .find((p) => p.type === 'application/sdp')
+        .content;
+      if (!sdp) {
+        logger.error({callId}, 'No SDP in multipart sdp');
+        return res.send(503);
+      }
+      const xml = req.payload.find((p) => p.type !== 'application/sdp');
+      const endPos = xml.content.indexOf('</recording>');
+      xml.content = endPos !== -1 ?
+        `${xml.content.substring(0, endPos + 12)}` :
+        xml.content;
+      logger.debug({callId, xml}, 'incoming call with SIPREC body');
+      req.locals = {...req.locals, sdp, siprec: true, xml};
+    }
+    else req.locals = {...req.locals, sdp: req.body};
+    next();
+  };
+
   const identifyAccount = async(req, res, next) => {
     try {
+      const {siprec, callId} = req.locals;
       const {wasOriginatedFromCarrier, getApplicationForDidAndCarrier} = req.srf.locals;
       const {
         fromCarrier,
@@ -132,8 +155,18 @@ module.exports = function(srf, logger) {
         }
         logger.debug({gateway}, 'identifyAccount: incoming call from gateway');
 
-        /* check for phone number level routing */
-        const sid = application_sid || await getApplicationForDidAndCarrier(req, gateway.voip_carrier_sid);
+        let sid;
+        if (siprec) {
+          if (!account.siprec_hook_sid) {
+            logger.info({callId}, 'identifyAccount: rejecting call because SIPREC hook has not been provisioned');
+            return res.send(404);
+          }
+          sid = account.siprec_hook_sid;
+        }
+        else {
+          /* check for phone number level routing */
+          sid = application_sid || await getApplicationForDidAndCarrier(req, gateway.voip_carrier_sid);
+        }
         req.locals = {
           originator: 'trunk',
           carrier: gateway.name,
@@ -296,6 +329,7 @@ module.exports = function(srf, logger) {
 
   return {
     initLocals,
+    handleSipRec,
     challengeDeviceCalls,
     identifyAccount,
     checkLimits

lib/utils.js

@@ -15,7 +15,13 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
   const from = req.getParsedHeader('from');
   const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
   const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
-  const srctOpts = srcIsUsingSrtp ? 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']
@@ -24,7 +30,7 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
     common,
     uas: {
       tag: from.params.tag,
-      mediaOpts: srctOpts
+      mediaOpts: srcOpts
     },
     uac: {
       tag: null,

package.json

@@ -1,6 +1,6 @@
 {
   "name": "sbc-inbound",
-  "version": "v0.7.5",
+  "version": "v0.7.6",
   "main": "app.js",
   "engines": {
     "node": ">= 12.0.0"
@@ -26,30 +26,30 @@
   },
   "dependencies": {
     "@jambonz/db-helpers": "^0.6.18",
-    "@jambonz/http-authenticator": "^0.2.0",
+    "@jambonz/http-authenticator": "^0.2.1",
     "@jambonz/http-health-check": "^0.0.1",
-    "@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",
-    "aws-sdk": "^2.1036.0",
+    "@jambonz/time-series": "^0.1.12",
+    "aws-sdk": "^2.1152.0",
     "bent": "^7.3.12",
     "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",
-    "express": "^4.17.3",
-    "husky": "^7.0.4",
-    "pino": "^7.4.1",
-    "rtpengine-client": "^0.2.0",
-    "verify-aws-sns-signature": "^0.0.6",
+    "drachtio-srf": "^4.5.1",
+    "express": "^4.18.1",
+    "pino": "^7.11.0",
+    "sdp-transform": "^2.14.1",
+    "uuid": "^8.3.2",
+    "verify-aws-sns-signature": "^0.0.7",
     "xml2js": "^0.4.23"
   },
   "devDependencies": {
-    "clear-module": "^4.1.1",
     "eslint": "^7.32.0",
     "eslint-plugin-promise": "^4.3.1",
     "nyc": "^15.1.0",
-    "tape": "^4.13.3"
+    "tape": "^4.15.1"
   }
 }

test/scenarios/uas.xml

@@ -124,7 +124,24 @@
   ]]>
 </send>
 
+<recv request="BYE">
+</recv>
+
+<send next="2">
+  <![CDATA[
+
+    SIP/2.0 200 OK
+    [last_Via:]
+    [last_From:]
+    [last_To:]
+    [last_Call-ID:]
+    [last_CSeq:]
+    Contact: <sip:[local_ip]:[local_port];transport=[transport]>
+    Content-Length: 0
+
+  ]]>
+</send>
+
 <label id="2"/>
 
 </scenario>
-