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/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,