2008/1/9, Johansson Olle E <[EMAIL PROTECTED]>: > > > 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.
Here (http://www3.tools.ietf.org/html/draft-ietf-sip-outbound-11#page-11 ยง3.5.2) using STUN technique is recommended. Do you foresee phone manufacturers to support this ? /O > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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