[asterisk-users] How to check if a SIP phone is forwardedwithout ringing it ?

[Johansson Olle E]

9 jan 2008 kl. 02.48 skrev Raj Jain:

> This issue of phone vendors not supporting OPTIONS according to RFC  
> 3261
> often comes up on this list. Like Kevin Fleming said, an OPTIONS  
> request is
> supposed to be responded in the same way as an INVITE. Almost all  
> SIP phone
> vendors have construed OPTIONS as some kind of a keep-alive request,  
> which
> is wrong.
Which we do too, by the way. In worst case, maybe Asterisk has set  
this industry
standard.

OPTIONS is far to heavy in processing on the server side to be used  
for keep-alives. I'm  starting to see devices that use it for checking  
capabilities - the proper way. To do this properly, we will have to  
authenticate the OPTIONs request and match it with the proper peer/ 
user to get the proper codec settings, ACLs and such.

Since all versions of Asterisk use OPTIONs for NAT-keepalives, I'm a  
bit hesitant to fix this. It's a catch 22. I want to do it properly,  
but then the amount of processing for each OPTIONs request that we  
receive is going to be a bit too much. Maybe one could ask vendors to  
add a header to the  OPTIONs packet saying "this is just a keep-alive.  
Give me a 200 OK without any parsing and be happy, because I don't  
care about the reply."

Linksys has a setting and use NOTIFY for Keep-alives, which also is a  
poor solution, but at least something we can just give an error  
response to without a lot of processing. There was a proposal for  
PING, but it never got anywhere.

/O