0208 sip rtcsample asterisk pc2phone debug ok
看来把微软的RTC放到最后是明智的,否则就没有什么耐心去搞LINUX那些开源东西了。总算编译运行PASS,可以PC2PC,PC2PHONE,应该视频也没有什么问题。服务器是用的asterisk NOt LCS
参考资料:
用微软的实时通信API集成丰富的客户端通信
使用 Microsoft 实时通信 API 增强多客户端通信
RTC 客户端 API 用于可缩放的应用程序
在应用程序中集成 Windows 实时通信
Integrating Rich Client Communications with Microsoft Real-Time Communications API
Enhancing Rich Client Communications with the Microsoft* Real-Time Communications API
Integrating Windows Real-Time Communications into Applications
Media Support in the Microsoft Windows Real-Time Communications Client(英文)
Integrating Rich Client Communications with the Microsoft Real-Time Communications API(英文)
Microsoft\’s real-time communication (RTC) API encapsulates some remarkable PC based communication innovations. The RTC API enables developers to enhance the communication capabilities of any Microsoft Windows XP* based application. It provides the developer with the building blocks to add instant messaging, voice and video-conferencing, and application sharing/collaboration capabilities to their application.
期间出现一些错误信息:
wsprintf(szBuf,
"<provision key=\\"%s\\" name=\\"AVDConf_2\\">"
"<user uri=\\"%s\\" account=\\"\\" password=\\"\\" realm=\\"%s\\" />"
"<sipsrv addr=\\"%s\\" protocol=\\"%s\\" %s role=\\"proxy\\">"
"<session party=\\"first\\" type=\\"pc2pc\\" />"
"</sipsrv>"
"<sipsrv addr=\\"%s\\" protocol=\\"%s\\" %s role=\\"registrar\\"/>"
"</provision>",
szKey,
szURIBuf, szRealm,
szSIPIP, szTransport, bBasicAuth ? "auth=\\"basic\\"" : "",
szSIPIP, szTransport, bBasicAuth ? "auth=\\"basic\\"" : ""
);
where szTransport="UDP" bBasicAuth=0 and Sip server requires no
authentication. I turned on tracing and found that the app poped up a dialog
saying"logon failed! Err=0×80ee001a) when calling
IRTCClientProvisining::EnableProfile() method ( CreateProfile( ) method
rerurned successfully.) . What might be pointed out is that I did not enable
the presence information . Now I am totally confused and wonder if you could
help figure out the reason why I couldn\’t logon on.
//
// MessageId: RTC_E_SIP_REQUEST_DESTINATION_ADDR_NOT_PRESENT
//
// MessageText:
//
// The Request destination address is not known
//
#define RTC_E_SIP_REQUEST_DESTINATION_ADDR_NOT_PRESENT ((HRESULT)0×80EE001AL)
//
// MessageId: RTC_E_DUPLICATE_REALM
//
// MessageText:
//
// Duplicate realm exists in an enabled profile
//
#define RTC_E_DUPLICATE_REALM ((HRESULT)0×80EE0043L)
//
// MessageId: RTC_E_UDP_NOT_SUPPORTED
//
// MessageText:
//
// This feature is not supported when one of the server in profile has UDP transport
//
#define RTC_E_UDP_NOT_SUPPORTED ((HRESULT)0×80EE007EL)
//
// MessageId: RTC_E_SIP_TCP_FAIL
//
// MessageText:
//
// Fails to make a TCP connection
//
#define RTC_E_SIP_TCP_FAIL ((HRESULT)0×80EE0067L)
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