Feature/minimal media anchoring (#36)

* initial WIP to remove freeswitch from media path when not recording or transcribing dial calls

* implement release-media and anchor-media operations

* mute/unmute now handled by rtpengine

* Dial: dtmf detection now based on SIP INFO events from sbcs and rtpengine

* add reason to gather action, bugfixes for transcribe and say

lib/session/call-session.js

@@ -926,6 +926,28 @@ class CallSession extends Emitter {
     };
   }
 
+  async releaseMediaToSBC(remoteSdp) {
+    assert(this.dlg && this.dlg.connected && this.ep && typeof remoteSdp === 'string');
+    await this.dlg.modify(remoteSdp, {
+      headers: {
+        'X-Reason': 'release-media'
+      }
+    });
+    this.ep.destroy()
+      .then(() => this.ep = null)
+      .catch((err) => this.logger.error({err}, 'CallSession:releaseMediaToSBC: Error destroying endpoint'));
+  }
+
+  async reAnchorMedia() {
+    assert(this.dlg && this.dlg.connected && !this.ep);
+    this.ep = await this.ms.createEndpoint({remoteSdp: this.dlg.remote.sdp});
+    await this.dlg.modify(this.ep.local.sdp, {
+      headers: {
+        'X-Reason': 'anchor-media'
+      }
+    });
+  }
+
   /**
    * Called any time call status changes.  This method both invokes the
    * call_status_hook callback as well as updates the realtime database