drachtio-srf 5.0.2 → 5.0.4

package/lib/digest-client.js

@@ -120,6 +120,8 @@ module.exports = class DigestClient {
         if (transport) proxy += `;transport=${transport}`;
         Object.assign(options, {proxy});
       }
+
+      options._socket = this.res.socket;
       this.agent.request(options, callback) ;
     }) ;
   }

package/lib/srf.js

@@ -298,6 +298,9 @@ class Srf extends Emitter {
 
     const __x = async(callback) => {
       try {
+        if (!req.has('Contact')) {
+          __fail(new Error('createUAS: Request is missing Contact header'), callback);
+        }
         const sdp = await generateSdp();
         debug({sdp}, `createUAS - generateSdp returned ${sdp}`);
         __send(sdp, callback);
@@ -549,9 +552,14 @@ class Srf extends Emitter {
                   return callback(null, {
                     sdp: res.body,
                     ack: (localSdp) => {
-                      return new Promise((resolve) => {
+                      return new Promise((resolve, reject) => {
                         ack({body: localSdp}) ;
 
+                        if (!res.has('Contact')) {
+                          debug('createUAC: no Contact header in response, returning 480');
+                          return reject(new Error('createUAC: no Contact header in response'));
+                        }
+
                         const dialog = new Dialog(this, 'uac', {req: req, res: res, auth: opts.auth}) ;
                         dialog.local.sdp = localSdp ;
                         this.addDialog(dialog) ;
@@ -574,6 +582,17 @@ class Srf extends Emitter {
 
               if ((200 === res.status && method === 'INVITE') ||
                   ((202 === res.status || 200 === res.status) && method === 'SUBSCRIBE')) {
+                if (!res.has('Contact')) {
+                  debug('createUAC: no Contact header in response, returning 500');
+                  // When UAC receive a 200 OK without Contact header, it is too late for UAC
+                  // to send CANCEL as it is already in the dialog.
+                  // UAC does not have contact header to send indialog request such as ACK or BYE.
+                  // The best way to handle this is to return 500 to the application.
+                  // Then leave the drachtio server to release the INVITE transaction as failed.
+                  const error = new SipError(500, 'No Contact header in response') ;
+                  error.res = res ;
+                  return callback(error);
+                }
                 const dialog = new Dialog(this, 'uac', {req: req, res: res, auth: opts.auth}) ;
                 if (req._dialogState) {
                   dialog.stateEmitter = {

package/package.json

@@ -1,6 +1,6 @@
 {
   "name": "drachtio-srf",
-  "version": "5.0.2",
+  "version": "5.0.4",
   "description": "drachtio signaling resource framework",
   "main": "lib/srf.js",
   "types": "lib/@types/index.d.ts",

package/test/b2b.js

@@ -19,6 +19,34 @@ test('B2B', (t) => {
   });
 
   Promise.resolve()
+    // B2BUA with PRACK
+    .then(() => {
+      debug('starting sipp');
+      return b2b.expectSuccess('sip:sipp-uas-prack', {
+        proxyResponseHeaders: [
+          'all'
+        ],
+        responseHeaders: {
+          'Contact': 'sip:foo@localhost'
+        }
+      });
+    })
+    .then(() => {
+      debug('start sipp...');
+      return sippUac('uac-prack.xml');
+    })
+    .then(() => {
+      return t.pass('b2b handles PRACK for both UAS and UAC');
+    })
+    .then(() => {
+      b2b.disconnect();
+      return new Promise((resolve, reject) => {
+        setTimeout(() => {
+          b2b = new B2b();
+          resolve();
+        }, 100);
+      });
+    })
     // INVITE with no SDP
     .then(() => {
       debug('starting sipp');
@@ -186,6 +214,50 @@ test('B2B', (t) => {
       });
     })
 
+    // reject no contact headers in sip request
+    .then(() => {
+      debug('starting sipp');
+      return b2b.expectFailure('sip:sipp-uas-404', 404, 400);
+    })
+    .then(() => {
+      debug('start sipp...');
+      return sippUac('uac-expect-400-no-contact-header.xml');
+    })
+    .then(() => {
+      return t.pass('reject if no contact headers in request');
+    })
+    .then(() => {
+      b2b.disconnect();
+      return new Promise((resolve, reject) => {
+        setTimeout(() => {
+          b2b = new B2b();
+          resolve();
+        }, 100);
+      });
+    })
+
+    // reject no contact headers in sip response
+    .then(() => {
+      debug('starting sipp');
+      return b2b.expectFailure('sipp-uas-200-ok-no-contact-cancel', 500, 480);
+    })
+    .then(() => {
+      debug('start sipp...');
+      return sippUac('uac-expect-480.xml');
+    })
+    .then(() => {
+      return t.pass('reject if no contact headers in response');
+    })
+    .then(() => {
+      b2b.disconnect();
+      return new Promise((resolve, reject) => {
+        setTimeout(() => {
+          b2b = new B2b();
+          resolve();
+        }, 100);
+      });
+    })
+
     // pass headers on response
     .then(() => {
       debug('starting sipp');

package/test/docker-compose-testbed.yaml

@@ -194,3 +194,14 @@ services:
     networks:
       testbed:
         ipv4_address: 172.29.0.27
+
+  sipp-uas-200-ok-no-contact-cancel:
+    image: drachtio/sipp:latest
+    command: sipp -sf /tmp/uas-200-ok-no-contact-cancel.xml
+    container_name: sipp-uas-200-ok-no-contact-cancel
+    volumes:
+      - ./scenarios:/tmp
+    tty: true
+    networks:
+      testbed:
+        ipv4_address: 172.29.0.28

package/test/scenarios/uac-expect-400-no-contact-header.xml

@@ -0,0 +1,81 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+<!--                                                                    -->
+<!--                 Sipp default 'uac' scenario.                       -->
+<!--                                                                    -->
+
+<scenario name="Basic Sipstone UAC">
+  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->
+  <!-- generated by sipp. To do so, use [call_id] keyword.                -->
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE 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]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="400" rtd="true"/>
+
+  <!-- Packet lost can be simulated in any send/recv message by         -->
+  <!-- by adding the 'lost = "10"'. Value can be [1-100] percent.       -->
+  <send>
+    <![CDATA[
+
+      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
+      [last_Via]
+      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+

package/test/scenarios/uac-prack.xml

@@ -0,0 +1,128 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<scenario name="UAC with PRACK Support">
+  <!-- Send INVITE with 100rel support -->
+  <send retrans="500">
+    <![CDATA[
+
+      INVITE 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]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Supported: 100rel
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv response="100" optional="true">
+  </recv>
+
+  <!-- Receive 180 Ringing with reliability required -->
+  <!-- Save RSeq and CSeq values for PRACK -->
+  <recv response="180" rrs="true">
+    <action>
+      <ereg regexp="([0-9]+)"
+            search_in="hdr" 
+            header="RSeq:"
+            check_it="true"
+            assign_to="1"/>
+      <ereg regexp="([0-9]+) INVITE"
+            search_in="hdr"
+            header="CSeq:"
+            check_it="true"
+            assign_to="2"/>
+    </action>
+  </recv>
+
+  <!-- Send PRACK for the 180 response -->
+  <send>
+    <![CDATA[
+
+      PRACK 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]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 2 PRACK
+      RAck: [$1] [$2] INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Receive 200 OK for PRACK -->
+  <recv response="200">
+  </recv>
+
+  <!-- Receive 200 OK for INVITE -->
+  <recv response="200" rtd="true">
+  </recv>
+
+  <!-- Send ACK for 200 OK 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]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 1 ACK
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Pause in conversation -->
+  <pause milliseconds="1000"/>
+
+  <!-- Send BYE -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE 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]SIPpTag00[call_number]
+      To: [service] <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
+      Call-ID: [call_id]
+      CSeq: 3 BYE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: Performance Test
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- Receive 200 OK for BYE -->
+  <recv response="200" crlf="true">
+  </recv>
+
+  <!-- definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>

package/test/scenarios/uas-200-ok-no-contact-cancel.xml

@@ -0,0 +1,111 @@
+<?xml version="1.0" encoding="ISO-8859-1" ?>
+<!DOCTYPE scenario SYSTEM "sipp.dtd">
+
+<!-- This program is free software; you can redistribute it and/or      -->
+<!-- modify it under the terms of the GNU General Public License as     -->
+<!-- published by the Free Software Foundation; either version 2 of the -->
+<!-- License, or (at your option) any later version.                    -->
+<!--                                                                    -->
+<!-- This program is distributed in the hope that it will be useful,    -->
+<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->
+<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->
+<!-- GNU General Public License for more details.                       -->
+<!--                                                                    -->
+<!-- You should have received a copy of the GNU General Public License  -->
+<!-- along with this program; if not, write to the                      -->
+<!-- Free Software Foundation, Inc.,                                    -->
+<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->
+<!--                                                                    -->
+<!--                 Sipp default 'uas' scenario.                       -->
+<!--                                                                    -->
+
+<scenario name="Basic UAS responder">
+  <!-- By adding rrs="true" (Record Route Sets), the route sets         -->
+  <!-- are saved and used for following messages sent. Useful to test   -->
+  <!-- against stateful SIP proxies/B2BUAs.                             -->
+  <recv request="INVITE" crlf="true">
+    <action>
+      <ereg regexp=".*" search_in="hdr" header="Via" check_it="false" assign_to="1"/>
+      <ereg regexp=".*" search_in="hdr" header="CSeq" check_it="false" assign_to="2"/>
+    </action>
+  </recv>
+
+  <!-- The '[last_*]' keyword is replaced automatically by the          -->
+  <!-- specified header if it was present in the last message received  -->
+  <!-- (except if it was a retransmission). If the header was not       -->
+  <!-- present or if no message has been received, the '[last_*]'       -->
+  <!-- keyword is discarded, and all bytes until the end of the line    -->
+  <!-- are also discarded.                                              -->
+  <!--                                                                  -->
+  <!-- If the specified header was present several times in the         -->
+  <!-- message, all occurrences are concatenated (CRLF separated)       -->
+  <!-- to be used in place of the '[last_*]' keyword.                   -->
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
+      s=-
+      c=IN IP[media_ip_type] [media_ip]
+      t=0 0
+      m=audio [media_port] RTP/AVP 0
+      a=rtpmap:0 PCMU/8000
+
+    ]]>
+  </send>
+
+  <recv request="CANCEL">
+  </recv>
+
+  <send>
+    <![CDATA[
+
+      SIP/2.0 200 OK
+      [last_Via:]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      [last_CSeq:]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+
+  <send retrans="500">
+    <![CDATA[
+
+      SIP/2.0 487 Request Terminated
+      Via [$1]
+      [last_From:]
+      [last_To:];tag=[pid]SIPpTag01[call_number]
+      [last_Call-ID:]
+      CSeq [$2]
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <recv request="ACK"
+        rtd="true"
+        crlf="true">
+  </recv>
+
+  <!-- definition of the response time repartition table (unit is ms)   -->
+  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
+
+  <!-- definition of the call length repartition table (unit is ms)     -->
+  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
+
+</scenario>
+