add support for sending calls to a sip uri through a proxy

lib/call-session.js

@@ -211,6 +211,10 @@ class CallSession extends Emitter {
       }
       else if (this.req.locals.target === 'forward') {
         uris = [this.req.uri];
+        if (this.req.has('X-SIP-Proxy')) {
+          proxy = this.req.get('X-SIP-Proxy');
+          if (!proxy.startsWith('sip:') && !proxy.startsWith('sips:')) proxy = `sip:${proxy}`;
+        }
       }
       else if (teams) {
         const vmailParam = 'opaque=app:voicemail';
@@ -389,6 +393,7 @@ class CallSession extends Emitter {
               '-X-Preferred-From-Host',
               '-X-Jambonz-FS-UUID',
               '-X-Voip-Carrier-Sid',
+              '-X-SIP-Proxy'
             ],
             proxyResponseHeaders: [
               'all',

test/scenarios/uac-sip-uri-proxy.xml

@@ -0,0 +1,120 @@
+<?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 'uac' scenario with pcap (rtp) play           -->
+<!--                                                                    -->
+
+<scenario name="UAC with media">
+  <!-- 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:john@foo.bar 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:john@foo.bar>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: sip:sipp@[local_ip]:[local_port]
+      Max-Forwards: 70
+      Subject: uac-sip-uri-proxy
+      X-Account-Sid: ed649e33-e771-403a-8c99-1780eabbc803
+      X-Call-Sid: fff49e33-e771-403a-8c99-1780eabbc803
+      X-Jambonz-Routing: sip
+      X-SIP-Proxy: 172.39.0.20:5060
+      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">
+  </recv>
+
+  <!-- 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:john@foo.bar SIP/2.0
+      [last_Via:]
+      [last_From:]
+      To: <sip:sip:john@jambonz.org>[peer_tag_param]
+      [last_Call-ID:]
+      CSeq: 2 ACK
+      Subject: 
+      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:john@foo.bar SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      [last_From:]
+      [last_To:]
+      [last_Call-ID:]
+      CSeq: 3 BYE
+      Max-Forwards: 70
+      Subject: uac-sip-uri-proxy
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <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>

test/sip-tests.js

@@ -84,6 +84,10 @@ test('sbc-outbound tests', async(t) => {
     await sippUac('uac-sip-uri-auth-success.xml');
     t.pass('successfully connected to sip uri that requires auth');
 
+    /* invite to sipUri through proxy */
+    await sippUac('uac-sip-uri-proxy.xml');
+    t.pass('successfully connected to sip uri through proxy');
+
     // re-rack test data
     execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/jambones-sql.sql`);
     execSync(`mysql -h 127.0.0.1 -u root  --protocol=tcp -D jambones_test < ${__dirname}/db/populate-test-data5.sql`);
@@ -96,7 +100,7 @@ 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 === 8, 'wrote 8 cdrs');
+    t.ok(res.total === 9, 'wrote 9 cdrs');
 
     srf.disconnect();
   } catch (err) {