Feature/sip info dtmf (#44)

* 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

lib/call-session.js

@@ -3,7 +3,6 @@ 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');
-const { v4: uuidv4 } = require('uuid');
 const debug = require('debug')('jambonz:sbc-inbound');
 const MS_TEAMS_USER_AGENT = 'Microsoft.PSTNHub.SIPProxy';
 const MS_TEAMS_SIP_ENDPOINT = 'sip.pstnhub.microsoft.com';
@@ -20,7 +19,7 @@ const createBLegFromHeader = (req) => {
 };
 
 const createSiprecBody = (headers, sdp, type, content) => {
-  const sep = uuidv4();
+  const sep = 'uniqueBoundary';
   headers['Content-Type'] = `multipart/mixed;boundary="${sep}"`;
   return `--${sep}\r
 Content-Type: application/sdp\r
@@ -51,6 +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() {
@@ -61,6 +61,10 @@ class CallSession extends Emitter {
     return this.srf.locals.privateSipAddress;
   }
 
+  get isMediaReleased() {
+    return this._mediaReleased;
+  }
+
   async connect() {
     const {sdp} = this.req.locals;
     this.logger.info('inbound call accepted for routing');
@@ -80,6 +84,7 @@ class CallSession extends Emitter {
       unblockMedia,
       blockDTMF,
       unblockDTMF,
+      playDTMF,
       subscribeDTMF,
       unsubscribeDTMF,
       subscribeRequest,
@@ -93,6 +98,7 @@ 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;
@@ -487,6 +493,7 @@ Duration=${payload.duration} `
         this.logger.info({response}, `got a reinvite from FS to ${reason}`);
         sdp = dlg.other.remote.sdp;
         answerMedia.flags = ['asymmetric', 'port latching'];
+        this._mediaReleased = 'release-media' === reason;
       }
       else {
         sdp = await dlg.other.modify(response.sdp);
@@ -512,6 +519,7 @@ Duration=${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');
@@ -617,6 +625,40 @@ Duration=${payload.duration} `
           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'];
         const headers = {};

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,