1-1."Carrier Grade Voice over IP," p215-222
And I also review chapter 1~2 to prepare for themidterm of the class, VoIP Measurement
1-2. RFC3261 section 22
In order to learn the authentication of SIP, I study section 22. In the OpenSER configuration file, it has the implementation codes for the specification authentication. I am glad to find it. (It seems that I can "trace code"?!)
2.Implementation Tasks:
2-1. Learn how to update the MySQL table which is storing the mapping between username and phone number.
2-2.The SIP server can do the authentication work, but the users on the server cannot call others. The users can register and get the other one call him. I also catch the packets between the call flow. When the registered user calls out, no packet comes from the user agent.
(And I also tried two scenarios: using private IP and public IP. When I use public IP, I even can't register on the the SIP server with IP:163.22.20.154. The gnome package for VoIP,Ekiga said, "There is other program listening the port. You cannot make a SIP call.")
The set-up of SIP server seems so easy for others and it is so difficult for me. Why?(Work hours:over 16 hours this week...not including my co-worker's:wenjen)
==============================================
Aaron Solomon using FreeBSD 6.2 |
| 顯示詳細資料 | 11月27日 |
Dear Hsiao-Ting: I am puzzled by what you wrote in your report: "When I use public IP, I even can't register on the the SIP server with IP:163.22.20.154." Because you did not specify the UA you were using, I was unable to check that for you. Please try to write down the problem which you encountered, so that we can discuss how to solve it. The key to solve a complicated problem is always "Divide and Conquer". Try to isolate the problem. Try to find out whether the problem is on the server or on the client. After you divide the problem into many small pieces, you will find it easier to handle in the smaller scale. With best regards, Solomon Nov. 27 |
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