fix outbound info (#107)

* Update _onFeatureServerTransfer to fix REFER case

* Avoid remote re-invite by using port latching

* Add INFO listener and pretend we're a uas

---------

Co-authored-by: Matt Preskett <matt.preskett@netcall.com>

app.js

@@ -32,7 +32,10 @@ const JAMBONES_REDIS_SENTINELS = process.env.JAMBONES_REDIS_SENTINELS ? {
   }),
   ...(process.env.JAMBONES_REDIS_SENTINEL_USERNAME && {
     username: process.env.JAMBONES_REDIS_SENTINEL_USERNAME
-  })
+  }),
+  ...(process.env.JAMBONES_REDIS_SENTINEL_SENTINAL_PASSWORD && {
+    sentinelPassword: process.env.JAMBONES_REDIS_SENTINEL_SENTINAL_PASSWORD
+  }),
 } : null;
 
 const Srf = require('drachtio-srf');

lib/call-session.js

@@ -99,6 +99,10 @@ class CallSession extends Emitter {
     this._mediaReleased = false;
   }
 
+  get service_provider_sid() {
+    return this.req.locals.service_provider_sid;
+  }
+
   get account_sid() {
     return this.req.locals.account_sid;
   }
@@ -173,7 +177,7 @@ class CallSession extends Emitter {
     this.unsubscribe = unsubscribe;
     this.srsClients = [];
 
-    this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, teams);
+    this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, this.useWss || teams, false, teams);
     this.rtpEngineResource = {destroy: this.del.bind(null, this.rtpEngineOpts.common)};
     let proxy, uris;
     const mapGateways = new Map();
@@ -289,8 +293,8 @@ class CallSession extends Emitter {
                  * but it will require more significant changes and right now it seems
                  * like a rare use case -- encryption is usually an all or nothing requirement.
                  */
-                this.logger.info({u}, 'using SRTP for outbound call');
-                this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, true);
+                this.logger.info({u}, `using SRTP for outbound call, pad crypto: ${o.pad_crypto ? 'yes' : 'no'}`);
+                this.rtpEngineOpts = makeRtpEngineOpts(this.req, false, true, o.pad_crypto, true);
               }
             });
             this.logger.debug({uris, voip_carrier_sid}, 'selected outbound gateways for requested carrier');
@@ -348,9 +352,10 @@ class CallSession extends Emitter {
             this.logger.info({err}, 'Error retrieving iip key');
           }
         }
+        // INVITE request line and To header should be the same.
+        hdrs = {...hdrs, 'To': uri};
         if (gw) {
           this.logger.info({gw}, `sending INVITE to ${uri} via carrier ${gw.name}`);
-          hdrs = {...hdrs, 'To': uri};
           if (gw.diversion) {
             let div = gw.diversion;
             if (div.startsWith('+')) {
@@ -856,15 +861,17 @@ Duration=${payload.duration} `
         }
       }
       else {
-        const response = await dlg.other.request({
-          method: 'INFO',
-          headers: req.headers,
-          body: req.body
-        });
-        res.send(response.status, {
-          headers: response.headers,
-          body: response.body
+        const immutableHdrs = ['via', 'from', 'to', 'call-id', 'cseq', 'max-forwards', 'content-length'];
+        const headers = {};
+        Object.keys(req.headers).forEach((h) => {
+          if (!immutableHdrs.includes(h)) headers[h] = req.headers[h];
         });
+        const response = await dlg.other.request({method: 'INFO', headers, body: req.body});
+        const responseHeaders = {};
+        if (response.has('Content-Type')) {
+          Object.assign(responseHeaders, {'Content-Type': response.get('Content-Type')});
+        }
+        res.send(response.status, {headers: responseHeaders, body: response.body});
       }
     } catch (err) {
       this.logger.info({err}, `Error handing INFO request on ${dlg.type} leg`);
@@ -902,35 +909,38 @@ Duration=${payload.duration} `
         ...(req.has('X-Retain-Call-Sid') && {'X-Retain-Call-Sid': req.get('X-Retain-Call-Sid')}),
         ...(req.has('X-Account-Sid') && {'X-Account-Sid': req.get('X-Account-Sid')})
       };
-      const dlg = await this.srf.createUAC(referTo.uri, {localSdp: dlg.local.sdp, headers});
-      this.uas = dlg;
-      this.uas.other = this.uac;
-      this.uac.other = this.uas;
-      this.uas.on('modify', this._onReinvite.bind(this, this.uas));
-      this.uas.on('refer', this._onFeatureServerTransfer.bind(this, this.uas));
-      this.uas.on('destroy', () => {
+
+      const uac = await this.srf.createUAC(referTo.uri, {localSdp: dlg.local.sdp, headers});
+      this.uas = uac;
+      uac.type = 'uas';
+      uac.other = this.uac;
+      this.uac.other = uac;
+      uac.on('info', this._onInfo.bind(this, uac));
+      uac.on('modify', this._onReinvite.bind(this, uac));
+      uac.on('refer', this._onFeatureServerTransfer.bind(this, uac));
+      uac.on('destroy', () => {
         this.logger.info('call ended with normal termination');
         this.rtpEngineResource.destroy();
         this.activeCallIds.delete(this.req.get('Call-ID'));
         if (this.activeCallIds.size === 0) this.idleEmitter.emit('idle');
-        this.uas.other.destroy();
+        uac.other.destroy();
         this.srf.endSession(this.req);
       });
 
-      // modify rtpengine to stream to new feature server
-      let response = await this.offer(Object.assign(this.rtpEngineOpts.offer, {sdp: this.uas.remote.sdp}));
+      const opts = {
+        ...this.rtpEngineOpts.common,
+        'from-tag': this.rtpEngineOpts.uas.tag,
+        sdp: uac.remote.sdp,
+        flags: ['port latching']
+      };
+      const response = await this.offer(opts);
       if ('ok' !== response.result) {
-        throw new Error(`_onReinvite: rtpengine failed: offer: ${JSON.stringify(response)}`);
-      }
-      const sdp = await this.uas.other.modify(response.sdp);
-      const opts = Object.assign({sdp, 'to-tag': res.getParsedHeader('To').params.tag},
-        this.rtpEngineOpts.answer);
-      response = await this.answer(opts);
-      if ('ok' !== response.result) {
-        throw new Error(`_onReinvite: rtpengine failed: ${JSON.stringify(response)}`);
+        throw new Error(`_onFeatureServerTransfer: rtpengine offer failed: ${JSON.stringify(response)}`);
       }
+      dlg.destroy().catch(() => {});
       this.logger.info('successfully moved call to new feature server');
     } catch (err) {
+      res.send(488);
       this.logger.error(err, 'Error handling refer from feature server');
     }
   }

lib/utils.js

@@ -2,10 +2,16 @@ const rtpCharacteristics = require('../data/rtp-transcoding');
 const srtpCharacteristics = require('../data/srtp-transcoding');
 const debug = require('debug')('jambonz:sbc-outbound');
 
-function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, teams = false) {
+function makeRtpEngineOpts(req, srcIsUsingSrtp, dstIsUsingSrtp, padCrypto, teams) {
   const from = req.getParsedHeader('from');
   const rtpCopy = JSON.parse(JSON.stringify(rtpCharacteristics));
   const srtpCopy = JSON.parse(JSON.stringify(srtpCharacteristics));
+
+  if (padCrypto) {
+    srtpCopy['default'].flags.push('SDES-pad');
+    srtpCopy['teams'].flags.push('SDES-pad');
+  }
+
   const srtpOpts = teams ? srtpCopy['teams'] : srtpCopy['default'];
   const dstOpts = dstIsUsingSrtp ? srtpOpts : rtpCopy;
   const srcOpts = srcIsUsingSrtp ? srtpOpts : rtpCopy;

test/scenarios/uac-pcap-sip-routing-success.xml

@@ -0,0 +1,91 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAC with media">
+
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE sip:16173333456@172.39.0.20 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:16173333456@172.39.0.20>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
+      X-Call-Sid: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: sip
+      Subject: uac-pcap-carrier-success
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      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 8 101
+      a=rtpmap:8 PCMA/8000
+      a=rtpmap:101 telephone-event/8000
+      a=fmtp:101 0-11,16
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="200" rtd="true" crlf="true">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      ACK sip:sip:+16173333456@127.0.0.1 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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Play a pre-recorded PCAP file (RTP stream)                       -->
+  <nop>
+    <action>
+      <exec play_pcap_audio="pcap/g711a.pcap"/>
+    </action>
+  </nop>
+
+  <!-- Pause briefly      -->
+  <pause milliseconds="2000"/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE sip:sip:+16173333456@127.0.0.1 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:sip:+16173333456@127.0.0.1>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 BYE
+      Max-Forwards: 70
+      Subject: uac-pcap-carrier-success
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv response="200" crlf="true">
+  </recv>
+
+</scenario>

test/sip-tests.js

@@ -44,6 +44,10 @@ test('sbc-outbound tests', async(t) => {
     await sippUac('uac-pcap-carrier-success.xml');
     t.pass('successfully completed outbound call to sip trunk');
 
+    /* call to Sip URI with no lcr configured */
+    await sippUac('uac-pcap-sip-routing-success.xml');
+    t.pass('successfully completed outbound call to sip routing trunk');
+
     /* call to PSTN with no lcr configured */
     await sippUac('uac-pcap-inbound-carrier-success.xml');
     t.pass('successfully completed outbound call to sip trunk');
@@ -92,7 +96,7 @@ test('sbc-outbound tests', async(t) => {
 
     const res = await queryCdrs({account_sid: 'ed649e33-e771-403a-8c99-1780eabbc803'});
     console.log(`${res.total} cdrs: ${JSON.stringify(res)}`);
-    t.ok(res.total === 7, 'wrote 7 cdrs');
+    t.ok(res.total === 8, 'wrote 8 cdrs');
 
     srf.disconnect();
   } catch (err) {