Client on 4G can't get INVITE

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Choi
Posts: 5
Joined: Tue Mar 15, 2016 4:14 pm

Client on 4G can't get INVITE

Post by Choi »

1. Brekeke Product Name and Version: 3.5.3.0, Standard

2. Java version: 1.8.0_77

3. OS type and the version: Windows Server 2008

4. UA (phone), gateway or other hardware/software involved: Phone on 4G connection

5. Your problem: A client works fine with registration and a call on WiFi, but it can't get a call from others after switching into 4G.

Brekeke SIP Server is located in the public (behind NAT), and every client is outside of BSS - perhaps network Pattern5.


Following session log shows minor difference between wifi and 4G connection.

1. Call to a client on 4G from a client on WiFi: Time Out, 504, 98.189.75.178:28460, 30.36.50.34:6291
--> The client on 4G can't get "Invite" message
2. Call to a client on WiFi from a client on WiFI: Success, -1, 98.189.75.178:28460, 98.189.75.178:17065

Port for 4G is lesser than 10000, and for wifi it's greater than 10000. I guess the port for 4G is blocked on firewall. FYI, Our server opened the following ports suggested on your manual.
- UDP and TCP: 5060
- UDP: 10000 - 29999

Is there a port to be opened for 4G, or something else to be set on BSS?
Choi
Posts: 5
Joined: Tue Mar 15, 2016 4:14 pm

Post by Choi »

One more for your understanding.

Case A) Client1(wifi) --> Client2(4G) : Client2 can't get INVITE
Case B) Client2(4G) --> Client1(wifi) : Client1 can get INVITE and talk together
Case C) Client1(wifi) <--> Client2(4G) : works fine at any direction

When makes a call,
- On Case A, Client1 uses "Source IP Address" but Client2 uses ip of "Contact URI"
- On Case B, Client 1 & 2 use "Source IP Address"
- On Case C, Client 1 & 2 use "Source IP Address"

* These are found on Session Logs
Choi
Posts: 5
Joined: Tue Mar 15, 2016 4:14 pm

Post by Choi »

It seems getting closer .... . This is my solution though I don't make myself understood fully.

1. Add a dial plan
- matching patterns : $request = ^REGISTER
- deploy pattern : $continue = true, &register.contact.remote = true, &register.contact.nat = true

2. Add a variable on Congiruation>Advanced
- net.sip.fixed.addrport.uas = true


Anyone, does it make sense? :D
taitan
Posts: 237
Joined: Sat Mar 15, 2008 10:39 pm

Post by taitan »

Which SIP client products are you using?
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