diff --git a/lib/call-session.js b/lib/call-session.js
index 7a82c5e..56622e7 100644
--- a/lib/call-session.js
+++ b/lib/call-session.js
@@ -69,6 +69,8 @@ class CallSession extends Emitter {
     this.performLcr = performLcr;
     this.lookupCarrierBySid = lookupCarrierBySid;
     this.lookupSipGatewaysByCarrier = lookupSipGatewaysByCarrier;
+
+    this._mediaReleased = false;
   }
 
   get account_sid() {
@@ -79,6 +81,10 @@ class CallSession extends Emitter {
     return this.srf.locals.privateSipAddress;
   }
 
+  get isMediaReleased() {
+    return this._mediaReleased;
+  }
+
   async connect() {
     const teams = this.teams = this.req.locals.target === 'teams';
     const engine = this.srf.locals.getRtpEngine();
@@ -95,6 +101,7 @@ class CallSession extends Emitter {
       unblockMedia,
       blockDTMF,
       unblockDTMF,
+      playDTMF,
       subscribeDTMF,
       unsubscribeDTMF,
       subscribeRequest,
@@ -109,6 +116,7 @@ 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;
@@ -560,6 +568,7 @@ Duration=${payload.duration} `
         this.logger.info(`got a reinvite from FS to ${reason}`);
         sdp = dlg.other.remote.sdp;
         answerMedia.flags = ['asymmetric', 'port latching'];
+        this._mediaReleased = 'release-media' === reason;
       }
       else {
         sdp = await dlg.other.modify(response.sdp);
@@ -586,6 +595,7 @@ Duration=${payload.duration} `
 
   async _onInfo(dlg, req, res) {
     try {
+      const contentType = req.get('Content-Type');
       if (dlg.type === 'uas' && req.has('X-Reason')) {
         const toTag = this.rtpEngineOpts.uac.tag;
         const reason = req.get('X-Reason');
@@ -688,6 +698,40 @@ Duration=${payload.duration} `
           res.send(succeeded ? 200 : 503);
         }
       }
+      else if (dlg.type === 'uac' && ['application/dtmf-relay', 'application/dtmf'].includes(contentType)) {
+        const arr = /Signal=\s*([1-9#*])/.exec(req.body);
+        if (!arr) {
+          this.logger.info({body: req.body}, '_onInfo: invalid INFO dtmf request');
+          throw new Error(`_onInfo: no dtmf in body for ${contentType}`);
+        }
+        const code = arr[1];
+        const arr2 = /Duration=\s*(\d+)/.exec(req.body);
+        const duration = arr2 ? arr2[1] : 250;
+
+        if (this.isMediaReleased) {
+          /* just relay on to the feature server */
+          this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, relaying to feature server');
+          this._onDTMF(dlg.other, {event: code, duration})
+            .catch((err) => this.logger.info({err}, 'Error relaying DTMF to feature server'));
+          res.send(200);
+        }
+        else {
+          /* else convert SIP INFO to RFC 2833 telephony events */
+          this.logger.info({code, duration}, 'got SIP INFO DTMF from caller, converting to RFC 2833');
+          const opts = {
+            ...this.rtpEngineOpts.common,
+            'from-tag': this.rtpEngineOpts.uac.tag,
+            code,
+            duration
+          };
+          const response = await this.playDTMF(opts);
+          if ('ok' !== response.result) {
+            this.logger.info({response}, `rtpengine playDTMF failed with ${JSON.stringify(response)}`);
+            throw new Error('rtpengine failed: answer');
+          }
+          res.send(200);
+        }
+      }
       else {
         const response = await dlg.other.request({
           method: 'INFO',
diff --git a/lib/utils.js b/lib/utils.js
index 209e97b..e42dbb4 100644
--- a/lib/utils.js
+++ b/lib/utils.js
@@ -7,6 +7,12 @@ function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
   const srtpOpts = teams ? srtpCharacteristics['teams'] : srtpCharacteristics['default'];
   const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCharacteristics;
   const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCharacteristics;
+
+  /* webrtc clients (e.g. sipjs) send DMTF via SIP INFO */
+  if ((srcIsUsingSrtp || dstIsUsingSrtp) && !teams) {
+    dstOpts.flags.push('inject DTMF');
+    srcOpts.flags.push('inject DTMF');
+  }
   const common = {
     'call-id': req.get('Call-ID'),
     'replace': ['origin', 'session-connection']