fix rtpengine re-invite inactive sdp

lib/freeswitch-call-handler.js

@@ -63,13 +63,13 @@ const handler = (req, res) => {
  * @param {*} sdp SDP offered by Freeswitch in leg2 INVITE
  * @param {*} res res SIP Response object
  */
-function createSdpForResponse(sessionId, sdp, res) {
+function createSdpForResponse(opts, sdp, res) {
   return new Promise((resolve, reject) => {
-    client.get(sessionId, (err, result) => {
+    client.get(opts.sessionId, (err, result) => {
       if (err) {
         return reject(err);
       }
-      resolve(payloadCombiner(sdp, result));
+      resolve(payloadCombiner(sdp, result, opts.sdp1, opts.sdp2));
     });
   });
 }
@@ -93,7 +93,7 @@ function handleIncomingSiprecInvite(req, res, opts) {
         passProvisionalResponses: false,
         headers,
         localSdpB: opts.sdp1,
-        localSdpA: createSdpForResponse.bind(null, opts.sessionId)
+        localSdpA: createSdpForResponse.bind(null, opts)
       };
 
       return srf.createB2BUA(req, res, fsUri, callOpts);

lib/payload-combiner.js

@@ -1,10 +1,12 @@
 const transform = require('sdp-transform');
 const debug = require('debug')('drachtio:siprec-recording-server');
 
-module.exports = function(sdp1, sdp2) {
+module.exports = function(sdp1, sdp2, originalSdp1, originalSdp2) {
   const sdpObj = [];
   sdpObj.push(transform.parse(sdp1));
   sdpObj.push(transform.parse(sdp2));
+  const originalParsedSdp1 = transform.parse(originalSdp1);
+  const originalParsedSdp2 = transform.parse(originalSdp2);
 
   //const arr1 = /^([^]+)(c=[^]+)t=[^]+(m=[^]+?)(a=[^]+)$/.exec(sdp1) ;
   //const arr2 = /^([^]+)(c=[^]+)t=[^]+(m=[^]+?)(a=[^]+)$/.exec(sdp2) ;
@@ -23,13 +25,23 @@ module.exports = function(sdp1, sdp2) {
 
   if (!sdpObj[0].media[0].label) sdpObj[0].media[0].label = 1;
   if (!sdpObj[1].media[0].label) sdpObj[1].media[0].label = 2;
+  // Update SDP Direction
+  if (originalParsedSdp1.media[0].direction === 'inactive') {
+    sdpObj[0].media[0].direction = 'inactive';
+  } else {
+    sdpObj[0].media[0].direction = 'recvonly';
+  }
+  if (originalParsedSdp2.media[0].direction === 'inactive') {
+    sdpObj[1].media[0].direction = 'inactive';
+  } else {
+    sdpObj[1].media[0].direction = 'recvonly';
+  }
 
   //const aLabel = sdp1.includes('a=label:') ? '' : 'a=label:1\r\n';
   //const bLabel = sdp2.includes('a=label:') ? '' : 'a=label:2\r\n';
 
   sdpObj[0].media = sdpObj[0].media.concat(sdpObj[1].media);
   const combinedSdp = transform.write(sdpObj[0])
-    .replace(/a=sendonly\r\n/g, '')
     .replace(/a=direction:both\r\n/g, '');
 
   debug(`combined ${combinedSdp}`);

lib/rtpengine-call-handler.js

@@ -7,6 +7,7 @@ const debug = require('debug')('drachtio:siprec-recording-server');
 module.exports = (req, res) => {
   const callid = req.get('Call-ID');
   const from = req.getParsedHeader('From');
+  const totag = v4();
   const logger = req.srf.locals.logger.child({callid});
   const opts = {
     req,
@@ -22,21 +23,51 @@ module.exports = (req, res) => {
   const rtpEngine = getAvailableRtpengine();
 
   parseSiprecPayload(opts)
-    .then(allocateEndpoint.bind(null, 'caller', rtpEngine))
-    .then(allocateEndpoint.bind(null, 'callee', rtpEngine))
+    .then(allocateEndpoint.bind(null, 'caller', rtpEngine, totag))
+    .then(allocateEndpoint.bind(null, 'callee', rtpEngine, totag))
     .then(respondToInvite)
     .then((dlg) => {
       logger.info(`call connected successfully, using rtpengine at ${JSON.stringify(rtpEngine.remote)}`);
+      dlg.on('modify', _onReinvite.bind(null, rtpEngine, logger, totag));
       return dlg.on('destroy', onCallEnd.bind(null, rtpEngine, opts));
     })
     .catch((err) => {
       logger.error(`Error connecting call: ${err}`);
     });
 };
 
-function allocateEndpoint(which, rtpEngine, opts) {
+function _onReinvite(rtpEngine, logger, totag, req, res) {
+  const callid = req.get('Call-ID');
+  const from = req.getParsedHeader('From');
+  const opts = {
+    req,
+    res,
+    logger,
+    callDetails: {
+      'call-id': callid,
+      'from-tag': from.params.tag,
+    }
+  };
+
+  parseSiprecPayload(opts)
+    .then(allocateEndpoint.bind(null, 'caller', rtpEngine, totag))
+    .then(allocateEndpoint.bind(null, 'callee', rtpEngine, totag))
+    .then((opts) => {
+      const body = constructSiprecPayload(opts.rtpengineCallerSdp, opts.rtpengineCalleeSdp, opts.sdp1, opts.sdp2);
+      return opts.res.send(200, {body});
+    })
+    .catch((err) => {
+      logger.error(`Error connecting call: ${err}`);
+    });
+
+  logger.info(`received SIPREC Re-invite: ${req.uri}`);
+}
+
+function allocateEndpoint(which, rtpEngine, totag, opts) {
+  // If audio is inactive, rtpengine will stop recording and there is no blank audio in record file.
+  const sdp = (which === 'caller' ? opts.sdp1 : opts.sdp2).replace(/a=inactive\r\n/g, 'a=sendonly\r\n');
   const args = Object.assign({}, opts.callDetails, {
-    'sdp': which === 'caller' ? opts.sdp1 : opts.sdp2,
+    sdp,
     'replace': ['origin', 'session-connection'],
     'transport protocol': 'RTP/AVP',
     'record call': 'yes',
@@ -47,7 +78,7 @@ function allocateEndpoint(which, rtpEngine, opts) {
     'rtcp-mux': ['accept'],
     'direction':  ['public', 'public'],
   });
-  if (which === 'callee') Object.assign(args, {'to-tag': v4()});
+  if (which === 'callee') Object.assign(args, {'to-tag': totag});
 
   debug(`callDetails: ${JSON.stringify(opts.callDetails)}`);
   debug(`rtpengine args for ${which}: ${JSON.stringify(args)}, sending to ${JSON.stringify(rtpEngine.remote)}`);
@@ -63,7 +94,7 @@ function allocateEndpoint(which, rtpEngine, opts) {
 
 function respondToInvite(opts) {
   const srf = opts.req.srf;
-  const payload = constructSiprecPayload(opts.rtpengineCallerSdp, opts.rtpengineCalleeSdp);
+  const payload = constructSiprecPayload(opts.rtpengineCallerSdp, opts.rtpengineCalleeSdp, opts.sdp1, opts.sdp2);
   return srf.createUAS(opts.req, opts.res, {localSdp: payload});
 }
 

test/call_tests.js

@@ -124,4 +124,4 @@ test('siprec with multiple rtpengines', (t) => {
       .on('error', (err) => {
         t.end(err, 'error connecting to drachtio');
       });
-}) ;
+}) ;

test/parser.js

@@ -8,7 +8,7 @@ function combineAndVerifyPayloads(filename, delimiter, t) {
     .then((data) => {
       const sdp = data.split('__split_here__');
       t.ok(sdp.length === 2, 'read two sdps');
-      const full = combinePayloads(sdp[0], sdp[1]);
+      const full = combinePayloads(sdp[0], sdp[1], sdp[0], sdp[1]);
       t.pass('combined payloads');
       t.end();
     })

test/scenarios/uac_siprec_pcap.xml

@@ -125,6 +125,98 @@
     </action>
   </nop>
 
+<send>
+  <![CDATA[
+
+  INVITE sip:drachtio:[remote_port] SIP/2.0
+  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+  From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+  To: sip:drachtio:[remote_port][peer_tag_param]
+  Call-ID: [call_id]
+  CSeq: 2 INVITE
+  Contact: <sip:[local_ip]:[local_port]>
+  Max-Forwards: 70
+  Subject: siprec Test
+  Content-Type: multipart/mixed; boundary=690DEB64
+  Content-Length: [len]
+
+  --690DEB64
+  Content-Type: application/sdp
+
+  v=0
+  o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+  s=-
+  c=IN IP[local_ip_type] [local_ip]
+  t=0 0
+  m=audio [auto_media_port] RTP/AVP 8
+  c=IN IP[local_ip_type] [local_ip]
+  a=rtpmap:8 PCMA/8000
+  a=direction:both
+  a=inactive
+  a=label:a_leg
+  m=audio 60000 RTP/AVP 8
+  c=IN IP[local_ip_type] [local_ip]
+  a=rtpmap:8 PCMA/8000
+  a=direction:both
+  a=inactive
+  a=label:b_leg
+
+  --690DEB64
+  Content-Type: application/rs-metadata
+  Content-Disposition: recording-session
+
+  <?xml version="1.0" encoding="UTF-8"?>
+    <recording xmlns='urn:ietf:params:xml:ns:recording:1'>
+      <datamode>complete</datamode>
+      <session session_id="XoM2pKdFR8C3eX33qJKcww==">
+        <sipSessionID>hvij5wK4Wx</sipSessionID>
+      </session>
+      <participant participant_id="omFyWlFETEyFz/KXNLe4iQ==">
+      <nameID aor="sip:6001@localhost"/>
+      </participant>
+      <participant participant_id="fLbZzLMgRXGp+F5w0hOt6w==">
+        <nameID aor="sip:101@localhost"/>
+      </participant>
+      <stream stream_id="1LiAtnDVR9m469/0L4bwCA==" session_id="XoM2pKdFR8C3eX33qJKcww==">
+      <label>1</label>
+      </stream>
+      <stream stream_id="WJgj5uxeRH+nwUah02dWqg==" session_id="XoM2pKdFR8C3eX33qJKcww==">
+      <label>2</label>
+      </stream>
+    </recording>
+
+  --690DEB64--
+
+  ]]>
+</send>
+
+<recv response="100" optional="true">
+</recv>
+
+<recv response="180" optional="true">
+</recv>
+
+<recv response="200" rtd="true" crlf="true">
+</recv>
+
+<!-- Send ACK for the 200 OK received from the re-INVITE -->
+<send>
+  <![CDATA[
+
+  ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+  Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+  From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag09[call_number]
+  To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+  Call-ID: [call_id]
+  CSeq: 2 ACK
+  Contact: sip:sipp@[local_ip]:[local_port]
+  Max-Forwards: 70
+  Subject: Performance Test
+  Content-Length: 0
+
+  ]]>
+</send>
+
   <!-- Pause 8 seconds, which is approximately the duration of the      -->
   <!-- PCAP file                                                        -->
   <pause milliseconds="3000"/>