Opensips CFG/Routing
$750-1500 USD
ชำระเงินเมื่อจัดส่ง
I need an OpenSIPS configuration that can effectively proxy all SIP data to a bunch of asterisk servers, and protecting them from SIP flooding attacks, and automatically removing a server from use and re-assigning the SIP clients to a new server when they register next.
1) Load a list of SIP Users/Group key/value pairs. (Mysql)
2) Load a list of Asterisk servers (Mysql)
3) Keep it’s own list of which Groups it has assigned to a particular server.
4) All incoming packets must be proxied out to the other side, rewriting the contact/from-domain to ensure that it stays between the calls
5) For incoming register packets, it should check the user against the list of sip users, and assign one of the asterisk servers that is not at full capacity (This is determined by the total number of endpoints currently registered on that server). Update a variable/array to reflect the server it has chosen, and then forward it (and all future data) to that server.
This is based on the Group, and all other extensions on the same Group should be forwarded to the same asterisk server.
6) If an asterisk server goes offline (Doesn't respond to Options or doesn't respond OK), then all phones are moved to a new server. (Not all to the same new server, but they have to be moved in a group)
7) Identify too many authentication/register packets per second and ignore IP
8) Identify too many messages per second and ignore IP
9) I’m guessing it has to be stateful so that it can remember what to rewrite the contact header back to for return messages.
10) All return messages should be sent back to originating port/ip for NAT traversal, and ignore the actual contact address.
หมายเลขโปรเจค: #7135925
เกี่ยวกับโปรเจกต์
freelancer 7 คน กำลังเสนอราคาในงานนี้ โดยมีราคาเฉลี่ยอยู่ที่ $1494
Hi, one thing that you must know, that i will able for work from 15.03.2015... Many tasks in your project makes it slower in implement.
Our team is specialized in VoIP development, VoIP and Asterisk especially is our main specialization. We have wide experience of building customized solutions, fully based on open source products. My bid is approxi เพิ่มเติม
engineer voip expert system on Asterisk, Freeswitch, Mera MVTS softswitch, Mera II Pro, Nex-Tone, Quintum Cisco we make your system settings according to the requirements within 30 days the price is US $ 3,000.0 เพิ่มเติม
