[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[Jan Janak]

On 13-10 17:11, John Todd wrote:
[...]
> SER is an excellent option as a front end to Asterisk.  It is a 
> "true" SIP proxy, whereas Asterisk is a hybrid, and SIP has not been 
> the primary focus of Asterisk development.  In fact, Asterisk's SIP 
> implementation is very limited (though it is extremely pragmatic.)
> 
> However, moving to SER does not solve any of the issues about the 
> proxy being behind a NAT, and I believe that SER will have the same 
> problems (though I could be wrong on this; I haven't experimented 
> with SER's ability to work from behind a NAT.)   SIP clients work 
> well enough behind NAT (most of them, anyway) but the servers are a 
> different story.

  SER can can become very helpful when it is run in the public
  internet and clients are behind NATs. For this case SER contains many
  "NAT helping functions" that can rewrite header fields, test
  if a client comes from behind a NAT, ping clients behind NATs (to keep
  the NAT binding open) and force RTP proxy usage when necesary.

  Along with RTP proxy SER can help any *symmetric* SIP user agent to
  get through NAT.

  (A symmetric SIP user agent is a user agent that uses the same source
  port for receiving signalling and media as for sending them. Vast
  majority of SIP user agents as of today is symmetric, including Windows
  Messenger, Cisco phones, Grandstream phone a.s.o.).

  There is also support for "proxy behind NAT", but it is mostly
  untested yet.

  Jan.