[solved] How to configure a STUN server for a sip peer?

Asterisk Asterisk Support
1

Solution:

By the time you get around to this, you are sunk

Instead, try forcing all traffic including RTP through the server

A certain client site has proven troublesome in terms of one way audio, and I’ve been asked to install a “STUN” server for a sip peer, and then do test calls on that peer to see if it cleared up the issue.

I checked google and the digium mailing list thoroughly but there seems to be a want of information

Questions:

  1. [solved ] Which STUN package to use?
    stun

  2. [solved ] How to install it?
    yum install stun

  3. How to configure it?
    (no hide nor hair of any options or docs)

  4. How to set the sip peer to “use” it so that it plays nice and passes two way audio as I have been asked.
    (complete lack of any or all information, scant written sources)

Thanks,

no man file

stun --help

Usage:
pk-command-not-found [OPTION…]

PackageKit Command Not Found

Help Options:
-h, --help Show help options

not a service

service stun status

Whats the deal?
stun: unrecognized service

2

Why don’t you use one of the existing, public stun servers ? BTW: I highly doubt using stun server is necessary to solve your problem.