[OpenSIPS-Users] Hairpin routing and Loop avoidance

[Iñaki Baz Castillo]

El Jueves, 25 de Junio de 2009, Steve Ames escribió:

> > This is a kwnown bug of Asterisk.
>
> Does that translate as I'm out of luck pending asterisk people fixing this,
> or can someone point me at a workaround or the right phrase to google up
> some additional info on the asterisk bug?

Take a look here:
  https://issues.asterisk.org/view.php?id=7403

Note however that even if there are some patches supposed to fix the problem, 
the fact is that it has been never checked that them work correctly (in my 
tests it just doesn't work at all). The "closed" status of the bug means 
*nothing*.

Some Asterisk's developers involved in fixing this issue have proved not no 
have enough knowledge of SIP (specially in some core sections of RFC 3261) as 
you can realize by reading some comments in the bug report. Sometimes they 
prefer to invent "how SIP works" instead of reading the RFC (i.e. CANCEL 
To_tag in parallel forking, comparision of SIP URI's...).

Asterisk is not friend of "advanced" SIP features as forking.
Also, it doesn't seem that Asterisk developers care a lot this kind of issues 
not related to SIP "just for phones".

I wouldn't expect this issue being "sometime" fixed. In order to send back an 
INVITE to Asterisk you need a B2BUA (so From/To tags and Call-ID are different 
in the request arriving to Asterisk).
A solution could be using SEMS as transparent B2BUA (without handling media). 
This is possible with an example module provided in default SEMS installation.
Other solution could be using Sippy B2BUA.

Note that I don't mean removing OpenSIPS. What I mean is:

  Asterisk ---------> OpenSIPS ----------> B2BUA
      <--------------------------------------


Regards.



-- 
Iñaki Baz Castillo <ibc at aliax.net>