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📄 scenario.cpp

📁 sip终端
💻 CPP
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              break;            default:              return(0);            }        } // end for    }  return(1);}int createIntegerTable(char * P_listeStr,                        unsigned int ** listeInteger,                        int * sizeOfList){  int nb=0;  char * ptr = P_listeStr;  char * ptr_prev = P_listeStr;  unsigned int current_int;  int sizeOf;  bool isANumber;   if(isWellFormed(P_listeStr, sizeOfList) == 1)    {      (*listeInteger) = new unsigned int[(*sizeOfList)];      while((*ptr) != ('\0'))        {          if((*ptr) == ',')            {              sscanf(ptr_prev, "%u", &current_int);              if (nb<(*sizeOfList))                (*listeInteger)[nb] = current_int;              nb++;              ptr_prev = ptr+1;            }          ptr++;        }      // Read the last      sscanf(ptr_prev, "%u", &current_int);       if (nb<(*sizeOfList))        (*listeInteger)[nb] = current_int;      nb++;      return(1);    }  return(0);}// TIP: to integrate an existing XML scenario, use the following sed line:// cat ../3pcc-controller-B.xml | sed -e 's/\"/\\\"/g' -e 's/\(.*\)/\"\1\\n\"/'char * default_scenario [] = {  /************* Default_scenario[0] ***************/(char *)"<?xml version=\"1.0\" encoding=\"ISO-8859-1\" ?>\n""<!DOCTYPE scenario SYSTEM \"sipp.dtd\">\n""\n""<!-- This program is free software; you can redistribute it and/or      -->\n""<!-- modify it under the terms of the GNU General Public License as     -->\n""<!-- published by the Free Software Foundation; either version 2 of the -->\n""<!-- License, or (at your option) any later version.                    -->\n""<!--                                                                    -->\n""<!-- This program is distributed in the hope that it will be useful,    -->\n""<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->\n""<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->\n""<!-- GNU General Public License for more details.                       -->\n""<!--                                                                    -->\n""<!-- You should have received a copy of the GNU General Public License  -->\n""<!-- along with this program; if not, write to the                      -->\n""<!-- Free Software Foundation, Inc.,                                    -->\n""<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->\n""<!--                                                                    -->\n""<!--                 Sipp default 'uac' scenario.                       -->\n""<!--                                                                    -->\n""\n""<scenario name=\"Basic Sipstone UAC\">\n""  <!-- In client mode (sipp placing calls), the Call-ID MUST be         -->\n""  <!-- generated by sipp. To do so, use [call_id] token.                -->\n""  <send retrans=\"500\">\n""    <![CDATA[\n""\n""      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0\n""      Via: SIP/2.0/[transport] [local_ip]:[local_port]\n""      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]\n""      To: sut <sip:[service]@[remote_ip]:[remote_port]>\n""      Call-ID: [call_id]\n""      CSeq: 1 INVITE\n""      Contact: sip:sipp@[local_ip]:[local_port]\n""      Max-Forwards: 70\n""      Subject: Performance Test\n""      Content-Type: application/sdp\n""      Content-Length: [len]\n""\n""      v=0\n""      o=user1 53655765 2353687637 IN IP4 127.0.0.1\n""      s=-\n""      t=0 0\n""      c=IN IP4 [media_ip]\n""      m=audio [media_port] RTP/AVP 0\n""      a=rtpmap:0 PCMU/8000\n""\n""    ]]>\n""  </send>\n""\n""  <recv response=\"100\"\n""        optional=\"true\">\n""  </recv>\n""\n""  <recv response=\"180\" optional=\"true\">\n""  </recv>\n""\n""  <!-- By adding rrs=\"true\" (Record Route Sets), the route sets         -->\n""  <!-- are saved and used for following messages sent. Useful to test   -->\n""  <!-- against stateful SIP proxies/B2BUAs.                             -->\n""  <recv response=\"200\" rtd=\"true\">\n""  </recv>\n""\n""  <!-- Packet lost can be simulated in any send/recv message by         -->\n""  <!-- by adding the 'lost = \"10\"'. Value can be [1-100] percent.       -->\n""  <send>\n""    <![CDATA[\n""\n""      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0\n""      Via: SIP/2.0/[transport] [local_ip]:[local_port]\n""      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]\n""      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]\n""      Call-ID: [call_id]\n""      CSeq: 1 ACK\n""      Contact: sip:sipp@[local_ip]:[local_port]\n""      Max-Forwards: 70\n""      Subject: Performance Test\n""      Content-Length: 0\n""\n""    ]]>\n""  </send>\n""\n""  <!-- This delay can be customized by the -d command-line option       -->\n""  <!-- or by adding a 'milliseconds = \"value\"' option here.             -->\n""  <pause/>\n""\n""  <!-- The 'crlf' option inserts a blank line in the statistics report. -->\n""  <send retrans=\"500\">\n""    <![CDATA[\n""\n""      BYE sip:[service]@[remote_ip]:[remote_port] SIP/2.0\n""      Via: SIP/2.0/[transport] [local_ip]:[local_port]\n""      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]\n""      To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]\n""      Call-ID: [call_id]\n""      CSeq: 2 BYE\n""      Contact: sip:sipp@[local_ip]:[local_port]\n""      Max-Forwards: 70\n""      Subject: Performance Test\n""      Content-Length: 0\n""\n""    ]]>\n""  </send>\n""\n""  <recv response=\"200\" crlf=\"true\">\n""  </recv>\n""\n""  <!-- definition of the response time repartition table (unit is ms)   -->\n""  <ResponseTimeRepartition value=\"10, 20, 30, 40, 50, 100, 150, 200\"/>\n""\n""  <!-- definition of the call length repartition table (unit is ms)     -->\n""  <CallLengthRepartition value=\"10, 50, 100, 500, 1000, 5000, 10000\"/>\n""\n""</scenario>\n""\n", /************* Default_scenario[1] ***************/(char *)"<?xml version=\"1.0\" encoding=\"ISO-8859-1\" ?>\n""<!DOCTYPE scenario SYSTEM \"sipp.dtd\">\n""\n""<!-- This program is free software; you can redistribute it and/or      -->\n""<!-- modify it under the terms of the GNU General Public License as     -->\n""<!-- published by the Free Software Foundation; either version 2 of the -->\n""<!-- License, or (at your option) any later version.                    -->\n""<!--                                                                    -->\n""<!-- This program is distributed in the hope that it will be useful,    -->\n""<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->\n""<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->\n""<!-- GNU General Public License for more details.                       -->\n""<!--                                                                    -->\n""<!-- You should have received a copy of the GNU General Public License  -->\n""<!-- along with this program; if not, write to the                      -->\n""<!-- Free Software Foundation, Inc.,                                    -->\n""<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->\n""<!--                                                                    -->\n""<!--                 Sipp default 'uas' scenario.                       -->\n""<!--                                                                    -->\n""\n""<scenario name=\"Basic UAS responder\">\n""  <!-- By adding rrs=\"true\" (Record Route Sets), the route sets         -->\n""  <!-- are saved and used for following messages sent. Useful to test   -->\n""  <!-- against stateful SIP proxies/B2BUAs.                             -->\n""  <recv request=\"INVITE\" crlf=\"true\">\n""  </recv>\n""\n""  <!-- The '[last_*]' keyword is replaced automatically by the          -->\n""  <!-- specified header if it was present in the last message received  -->\n""  <!-- (except if it was a retransmission). If the header was not       -->\n""  <!-- present or if no message has been received, the '[last_*]'       -->\n""  <!-- keyword is discarded, and all bytes until the end of the line    -->\n""  <!-- are also discarded.                                              -->\n""  <!--                                                                  -->\n""  <!-- If the specified header was present several times in the         -->\n""  <!-- message, all occurences are concatenated (CRLF seperated)        -->\n""  <!-- to be used in place of the '[last_*]' keyword.                   -->\n""\n""  <send>\n""    <![CDATA[\n""\n""      SIP/2.0 180 Ringing\n""      [last_Via:]\n""      [last_From:]\n""      [last_To:];tag=[call_number]\n""      [last_Call-ID:]\n""      [last_CSeq:]\n""      Contact: <sip:[local_ip]:[local_port];transport=[transport]>\n""      Content-Length: 0\n""\n""    ]]>\n""  </send>\n""\n""  <send retrans=\"500\">\n""    <![CDATA[\n""\n""      SIP/2.0 200 OK\n""      [last_Via:]\n""      [last_From:]\n""      [last_To:];tag=[call_number]\n""      [last_Call-ID:]\n""      [last_CSeq:]\n""      Contact: <sip:[local_ip]:[local_port];transport=[transport]>\n""      Content-Type: application/sdp\n""      Content-Length: [len]\n""\n""      v=0\n""      o=user1 53655765 2353687637 IN IP4 127.0.0.1\n""      s=-\n""      t=0 0\n""      c=IN IP4 [media_ip]\n""      m=audio [media_port] RTP/AVP 0\n""      a=rtpmap:0 PCMU/8000\n""\n""    ]]>\n""  </send>\n""\n""  <recv request=\"ACK\"\n""        optional=\"true\"\n""        rtd=\"true\"\n""        crlf=\"true\">\n""  </recv>\n""\n""  <recv request=\"BYE\">\n""  </recv>\n""\n""  <send>\n""    <![CDATA[\n""\n""      SIP/2.0 200 OK\n""      [last_Via:]\n""      [last_From:]\n""      [last_To:];tag=[call_number]\n""      [last_Call-ID:]\n""      [last_CSeq:]\n""      Contact: <sip:[local_ip]:[local_port];transport=[transport]>\n""      Content-Length: 0\n""\n""    ]]>\n""  </send>\n""\n""  <!-- Keep the call open for a while in case the 200 is lost to be     -->\n""  <!-- able to retransmit it if we receive the BYE again.               -->\n""  <pause milliseconds=\"4000\"/>\n""\n""\n""  <!-- definition of the response time repartition table (unit is ms)   -->\n""  <ResponseTimeRepartition value=\"10, 20, 30, 40, 50, 100, 150, 200\"/>\n""\n""  <!-- definition of the call length repartition table (unit is ms)     -->\n""  <CallLengthRepartition value=\"10, 50, 100, 500, 1000, 5000, 10000\"/>\n""\n""</scenario>\n""\n",/************* Default_scenario[2] ***************/(char *)"<?xml version=\"1.0\" encoding=\"ISO-8859-1\" ?>\n""<!DOCTYPE scenario SYSTEM \"sipp.dtd\">\n""\n""<!-- This program is free software; you can redistribute it and/or      -->\n""<!-- modify it under the terms of the GNU General Public License as     -->\n""<!-- published by the Free Software Foundation; either version 2 of the -->\n""<!-- License, or (at your option) any later version.                    -->\n""<!--                                                                    -->\n""<!-- This program is distributed in the hope that it will be useful,    -->\n""<!-- but WITHOUT ANY WARRANTY; without even the implied warranty of     -->\n""<!-- MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the      -->\n""<!-- GNU General Public License for more details.                       -->\n""<!--                                                                    -->\n""<!-- You should have received a copy of the GNU General Public License  -->\n""<!-- along with this program; if not, write to the                      -->\n""<!-- Free Software Foundation, Inc.,                                    -->\n""<!-- 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA             -->\n""<!--                                                                    -->\n""<!--                 Sipp default 'regexp client' scenario.             -->\n""<!--                                                                    -->\n""\n""<scenario name=\"Client with regexp scenario\">\n""  <send retrans=\"500\">\n""    <![CDATA[\n""\n""      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0\n""      Via: SIP/2.0/[transport] [local_ip]:[local_port]\n""      From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]\n""      To: sut <sip:[service]@[remote_ip]:[remote_port]>\n""      Call-ID: [call_id]\n""      CSeq: 1 INVITE\n""      Contact: sip:sipp@[local_ip]:[local_port]\n""      Max-Forwards: 70\n""      Subject: Performance Test\n""      Content-Type: application/sdp\n""      Content-Length: [len]\n""\n""      v=0\n""      o=user1 53655765 2353687637 IN IP4 127.0.0.1\n""      s=-\n""      t=0 0\n""      c=IN IP4 [media_ip]\n""      m=audio [media_port] RTP/AVP 0\n""      a=rtpmap:0 PCMU/8000\n""\n""    ]]>\n""  </send>\n""\n""  <recv response=\"100\"\n""        optional=\"true\">\n""  </recv>\n""\n""  <recv response=\"180\" optional=\"true\">\n""  </recv>\n""\n""  <recv response=\"200\" start_rtd=\"true\">\n""    <!-- Definition of regexp in the action tag. The regexp must follow -->\n""    <!-- the Posix Extended standard (POSIX 1003.2), see:               -->\n""    <!--                                                                -->\n""    <!--   http://www.opengroup.org/onlinepubs/007908799/xbd/re.html    -->\n""    <!--                                                                -->\n""    <!-- regexp    : Contain the regexp to use for matching the         -->\n""    <!--             received message                                   -->\n""    <!--             MANDATORY                                          -->\n""    <!-- search_in : msg (try to match against the entire message)      -->\n"

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