fix issue when sip uri has a question mark in the user part of sip uri (

#314) * fix issue when sip uri has a question mark in the user part of sip uri * add test case
drachtio · Nov 6, 2023 · 695b370 · 695b370
1 parent 2c397d3
commit 695b370
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diff --git a/src/drachtio.cpp b/src/drachtio.cpp
@@ -372,6 +372,10 @@ namespace drachtio {
             // no go: if we can't decode it then we have an invalid input
             return false ;
         }
+        if( nullptr == url->url_user && std::string::npos != uri.find("@")) {
+          DR_LOG(log_info) << "normalizeSipUri: invalid uri, user part contains invalid chars:" << uri ;
+          return false;
+        }
 
         /* we allow applications to just give us a phone number sometimes, and that ends up parsed into the host portion with no scheme */
         if( NULL == url->url_scheme && NULL == url->url_user && NULL != url->url_host ) {

diff --git a/test/scenarios/uac-invalid-uri-user.xml b/test/scenarios/uac-invalid-uri-user.xml
@@ -0,0 +1,99 @@
+<?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:-TXDOTFILE-FILEREVIEWCOV-INSINT%26FIRST60DAYS-SENTAOT-1STATTEMPTMADE-DCSDL-R/SREGOWNER-REVIEWISO%26LPR%26CARFAX-PR-R/SREGOWNER/DRIVERONFILE-RS/NIONFILE-IVRECOVER-IASENT-INSINTCOVREVSENT***PEND-INSINTCOV-AOT-EST-PAYORDENY?-CLOSE@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      Max-Forwards: 69
+      To: <sip:-TXDOTFILE-FILEREVIEWCOV-INSINT%26FIRST60DAYS-SENTAOT-1STATTEMPTMADE-DCSDL-R/SREGOWNER-REVIEWISO%26LPR%26CARFAX-PR-R/SREGOWNER/DRIVERONFILE-RS/NIONFILE-IVRECOVER-IASENT-INSINTCOVREVSENT***PEND-INSINTCOV-AOT-EST-PAYORDENY?-CLOSE@[remote_ip]:[remote_port]>
+      From: <sip:[email protected]>;tag=c11ralrakt
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      X-Nextiva-One-Session: lax4.int.voipdnsservers.com
+      X-Nextiva-Corp-Acct-Number: 3681069
+      X-Nextiva-User-UUID: 904baa30-d2db-11eb-a8c8-005056a33d5a
+      Allow-Events: conference,talk,hold
+      Contact: <sip:[email protected];transport=ws;ob>
+      Session-Expires: 90;refresher=uac
+      Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
+      Supported: timer,ice,replaces,outbound
+      User-Agent: JsSIP/3.20.0 DesktopApp/1.15.2 NextivaBrowserSoftphone/5.29.2
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 2868383482636261453 2 IN IP4 127.0.0.1
+      s=-
+      t=0 0
+      a=group:BUNDLE 0 1
+      a=extmap-allow-mixed
+      a=msid-semantic: WMS 8feaf12e-f910-403c-b695-21f9d52358c5
+      m=audio 29982 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
+      c=IN IP4 165.22.12.254
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="500" rtd="true" rrs="true">
+  </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 [next_url] 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
+      [routes]
+      Max-Forwards: 70
+      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>
diff --git a/test/scenarios/uac-long-uri.xml b/test/scenarios/uac-long-uri.xml
@@ -0,0 +1,116 @@
+<?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:901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068@[remote_ip]:[remote_port] SIP/2.0
+      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
+      From: <sip:sipp@[local_ip]:[local_port]>;tag=[pid]SIPpTag00[call_number]
+      To: <sip:901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068901262806890126280689012628068@[remote_ip]:[remote_port]>
+      Call-ID: [call_id]
+      CSeq: 1 INVITE
+      Contact: <sip:sipp@[local_ip]:[local_port]>
+      Max-Forwards: 70
+      Subject: very long sdp
+      Content-Type: application/sdp
+      Content-Length: [len]
+
+      v=0
+      o=- 2868383482636261453 2 IN IP4 127.0.0.1
+      s=-
+      t=0 0
+      a=group:BUNDLE 0 1
+      a=extmap-allow-mixed
+      a=msid-semantic: WMS 8feaf12e-f910-403c-b695-21f9d52358c5
+      m=audio 29982 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
+      c=IN IP4 165.22.12.254
+    ]]>
+  </send>
+
+  <recv response="100"
+        optional="true">
+  </recv>
+
+  <recv response="180" optional="true">
+  </recv>
+
+  <recv response="183" optional="true">
+  </recv>
+
+  <!-- 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 response="200" rtd="true" rrs="true">
+  </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 [next_url] 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
+      [routes]
+      Max-Forwards: 70
+      Content-Length: 0
+
+    ]]>
+  </send>
+
+  <!-- This delay can be customized by the -d command-line option       -->
+  <!-- or by adding a 'milliseconds = "value"' option here.             -->
+  <pause/>
+
+  <!-- The 'crlf' option inserts a blank line in the statistics report. -->
+  <send retrans="500">
+    <![CDATA[
+
+      BYE [next_url] 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 BYE
+      [routes]
+      Max-Forwards: 70
+      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>
diff --git a/test/test-fixtures-all.json b/test/test-fixtures-all.json
@@ -66,6 +66,12 @@
     "uac": {"name": "uac-long-sdp.xml", "target": "127.0.0.1:5090"},
     "message": "b2b: handles very long sdp"
   },
+  {
+    "server": {"config": "drachtio.conf.xml", "args": ["--memory-debug"]},
+    "script": {"name": "uas", "function": "b2b"},
+    "uac": {"name": "uac-invalid-uri-user.xml", "target": "127.0.0.1:5090"},
+    "message": "b2b: rejects invalid uri user part"
+  },
   {
     "server": {"config": "drachtio.conf.xml", "args": ["--memory-debug"]},
     "script": {"name": "uas", "function": "b2b"},

diff --git a/test/test-fixtures-uac-invalid-uri-user.json b/test/test-fixtures-uac-invalid-uri-user.json
@@ -0,0 +1,8 @@
+[ 
+  {
+    "server": {"config": "drachtio.conf.xml", "args": ["--memory-debug"]},
+    "script": {"name": "uas", "function": "b2b"},
+    "uac": {"name": "uac-invalid-uri-user.xml", "target": "127.0.0.1:5090"},
+    "message": "b2b: rejects invalid uri user part"
+  }
+]
diff --git a/test/test-fixtures-uac-long-uri.json b/test/test-fixtures-uac-long-uri.json
@@ -0,0 +1,9 @@
+[ 
+  {
+    "server": {"config": "drachtio.conf.xml", "args": ["--memory-debug"]},
+    "script": {"name": "uas", "function": "b2b"},
+    "uas": {"name": "uas-success.xml", "port": 5094, "transport": "tcp"},
+    "uac": {"name": "uac-long-uri.xml", "target": "127.0.0.1:5090"},
+    "message": "b2b: handles very long uri"
+  }
+]