invitemsgtest.cxx

SIP（Session Initiation Protocol）是由IETF定义

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static const char* const InviteMsgTest_cxx_Version =
    "$Id: InviteMsgTest.cxx,v 1.2.2.1 2003/02/12 02:56:39 sprajpat Exp $";

#include "global.h"
#include "SipAuthorization.hxx"
#include <cassert>
#include "cpLog.h"
#include "InviteMsg.hxx"
#include "Verify.hxx"
#include "Sptr.hxx"
#include "BaseUrl.hxx"
#include "SipUser.hxx"
#include "SipVia.hxx"
#include "SipSdp.hxx"
#include "SipDiversion.hxx"

using namespace Vocal;
    
//message sent by telecaster
char* msgtext1[] = { 
        "INVITE sip:5452@172.19.174.171:5060;user=phone SIP/2.0\r\n\
Via: SIP/2.0/UDP 172.19.175.150:5060;branch=78377a23f9fb8014efae61c726ec6e77.4\r\n\
Via: SIP/2.0/UDP 172.19.175.150:5060;branch=d1faef3fff22fbedaa794b8ea2abc7e5.2\r\n\
Via: SIP/2.0/UDP 172.19.174.170:5060\r\n\
From: \"5451\"<sip:5451@172.19.175.150>\r\n\
To: <sip:5452@172.19.175.150;user=phone>\r\n\
Call-ID: c4943000-75748-7058-2e323731@172.19.174.170\r\n\
CSeq: 101 INVITE\r\n\
Expires: 180\r\n\
Accept: application/sdp\r\n\
Record-Route: <sip:5452@172.19.175.150:5060;maddr=172.19.175.150>,<sip:5452@172.19.175.150:5060;maddr=172.19.175.150>\r\n\
Contact: <sip:5451@172.19.174.170:5060>\r\n\
Content-Length: 228\r\n\
User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled\r\n\
Content-Type: application/sdp\r\n\
\r\n\
v=0\r\n\
o=CiscoSystemsSIP-IPPhone-UserAgent 21803 10790 IN IP4 172.19.174.170\r\n\
s=SIP Call\r\n\
c=IN IP4 172.19.174.170\r\n\
t=0 0\r\n\
m=audio 28844 RTP/AVP 0 8 18 100\r\n\
a=rtpmap:0 pcmu/8000\r\n\
a=rtpmap:100 telephone-event/8000\r\n\
a=fmtp:100 0-11\r\n\
\r\n",
// this is the second message
"INVITE sip:7013000@172.19.175.70:5060;user=phone;phone-context=unknown SIP/2.0\r\n\
Via: SIP/2.0/UDP 172.19.175.71:5060;branch=c419c2f0a1077bdf16b34df6384becc5.1\r\n\
Via: SIP/2.0/UDP 172.19.175.226:52901\r\n\
From: \"7010300\"<sip:7010300@172.19.175.226>\r\n\
To: <sip:7013000@172.19.175.71;user=phone;phone-context=unknown>\r\n\
Call-ID: 5CF43908-1A820032-0-4D8E29AC@172.19.175.226\r\n\
CSeq: 101 INVITE\r\n\
Proxy-Authorization: Basic VovidaClassXSwitch\r\n\
Max-Forwards: 5\r\n\
Expires: 60\r\n\
Record-Route: <sip:7013000@172.19.175.71:5060;maddr=172.19.175.71>\r\n\
Contact: <sip:7010300@172.19.175.226:5060;user=phone>\r\n\
Content-Length: 138\r\n\
Timestamp: 732245189\r\n\
User-Agent: Cisco VoIP Gateway/ IOS 12.x/ SIP enabled\r\n\
Content-Type: application/sdp\r\n\
\r\n\
v=0\r\n\
o=CiscoSystemsSIP-GW-UserAgent 6459 1977 IN IP4 172.19.175.226\r\n\
s=SIP Call\r\n\
c=IN IP4 172.19.175.226\r\n\
t=0 0\r\n\
m=audio 20752 RTP/AVP 0\r\n\
\r\n",
"INVITE sip:[e-mail] SIP/2.0
TO :
 sip:[e-mail] ;   tag    = 1a1b1f1H33n
From   : \"J Rosenberg \\\\\\\"\" <sip:[e-mail]>
  ;
  tag = 98asjd8
CaLl-Id
 : 0ha0isndaksdj@10.1.1.1
cseq: 8
  INVITE
Via  : SIP  /   2.0 
 /UDP 
    135.180.130.133
Subject :
NewFangledHeader:   newfangled value
 more newfangled value
Content-Type: application/sdp
v:  SIP  / 2.0  / TCP     12.3.4.5   ;
  branch  =   9ikj8  ,
 SIP  /    2.0   / UDP  1.2.3.4   ; hidden   
m:\"Quoted string \\\"\\\"\" <sip:[e-mail]> ; newparam = newvalue ;
  secondparam = secondvalue  ; q = 0.33
  ,
 tel:[phone]

v=0
o=mhandley 29739 7272939 IN IP4 126.5.4.3
s= 
c=IN IP4 135.180.130.88
m=audio 49210 RTP/AVP 0 12
m=video 3227 RTP/AVP 31
a=rtpmap:31 LPC/8000

",
        "INVITE sip:5452@172.19.174.171;user=phone SIP/2.0\r\n\
Via: SIP/2.0/UDP 172.19.174.170:5060\r\n\
From: \"5451\"<sip:5451@172.19.175.150;user=phone>\r\n\
To: <sip:5452@172.19.175.150;user=phone>\r\n\
Call-ID: c4943000-75748-7058-2e323731@172.19.174.170\r\n\
CSeq: 101 INVITE\r\n\
Expires: 180\r\n\
Accept: application/sdp\r\n\
Record-Route: <sip:5452@172.19.175.150:5060;maddr=172.19.175.150>,<sip:5452@172.19.175.150:5060;maddr=172.19.175.150>\r\n\
Contact: <sip:5451@172.19.174.170:5060>\r\n\
Diversion: <sip:5451@172.19.174.170:5060>;reason=no-answer\r\n\
Content-Length: 228\r\n\
User-Agent: Cisco IP Phone/ Rev. 1/ SIP enabled\r\n\
Content-Type: application/sdp\r\n\
\r\n\
v=0\r\n\
o=CiscoSystemsSIP-IPPhone-UserAgent 21803 10790 IN IP4 172.19.174.170\r\n\
s=SIP Call\r\n\
c=IN IP4 172.19.174.170\r\n\
t=0 0\r\n\
m=audio 28844 RTP/AVP 0 8 18 100\r\n\
a=rtpmap:0 pcmu/8000\r\n\
a=rtpmap:100 telephone-event/8000\r\n\
a=fmtp:100 0-11\r\n\
\r\n"

};    
    
    
    
    
    

void testEncodeDecode()
{
    Data url;
    url = "sip:destination@192.168.5.12";
    Sptr <BaseUrl> baseurl = BaseUrl::decode(url);
    Sptr <SipUrl> tourl;
    tourl.dynamicCast(baseurl);
    
    int listenPort = 5060;
    int rtpPort = 3456;
    cpLog(LOG_DEBUG, "Completed the Url decoding");

    InviteMsg invite(tourl, listenPort, rtpPort);

    Data data = invite.encode();
    //cout << " before decoding " << data.logData() << endl;
    InviteMsg newInvite(data);

    Data newdata = newInvite.encode();
    
    InviteMsg testInvite(newdata);
    
    //cout << "New Invite Msg after encoding " << newdata.logData() << endl;
    
    
    test_verify (newInvite == testInvite);
    
    //test for telecaster msg

    Data tempInvite(msgtext1[0]);
    
    InviteMsg invite1(tempInvite);
    
    Data invStr1 = invite1.encode();
    
    InviteMsg testInvite1(invStr1);
    
     test_verify (invite1 == testInvite1); 
    
    //cout << "Invite msg - for telecaster msg BEFORE decoding: " << tempInvite.logData() << endl;
    //cout << "Invite msg - for telecaster msg AFTER decoding: " << invStr1.logData() << endl;
    
    //test for callgen msg
    Data tempInvite2(msgtext1[1]);
    
    InviteMsg invite2(tempInvite2);
    
    Data invStr2 = invite2.encode();
    
    InviteMsg testInvite2(invStr2);
    
    test_verify( testInvite2 == invite2);
    
    //cout << "Invite msg - for call gen msg BEFORE decoding: " << tempInvite2.logData() << endl;
    
     // cout << "Invite msg - for call gen msg AFTER decoding: " << invStr2.logData() << endl;
    
    
#if 0
    SipTo to(" <sip:7013000@172.19.175.71;user=phone;phone-context=unknown>");
    
    cout << to.encode().logData() << endl;
    
    SipRecordRoute rec("<sip:7013000@172.19.175.71:5060;maddr=172.19.175.71>");
    cout << rec.encode().logData() << endl;
    
    
    SipContact cont( "<sip:7010300@172.19.175.226:5060;user=phone>");
    cout << cont.encode().logData() << endl;
#endif


    {
	InviteMsg invite(msgtext1[2]);
	cout << invite.encode() << endl;
    }
}


char* msgtext[] = 
{ 
    "INVITE sip:5452@172.19.174.171:5060 SIP/2.0\r\n\
Via: SIP/2.0/UDP 172.19.175.150:5060;branch=78377a23f9fb8014efae61c726ec6e77.4\r\n\
Via: SIP/2.0/UDP 172.19.175.150:5060;branch=d1faef3fff22fbedaa794b8ea2abc7e5.2\r\n\
Via: SIP/2.0/UDP 172.19.174.170:5060\r\n\
From: \"5451\"<sip:5451@172.19.175.150>\r\n\
To: <sip:5452@172.19.175.150>\r\n\
Call-ID: c4943000-75748-7058-2e323731@172.19.174.170\r\n\
CSeq: 101 INVITE\r\n\
Expires: 180\r\n\
Accept: application/sdp\r\n\