Great doc! Some short notes: *This is broken. Cisco is the most well known vendor to make this mistake.
These implementations should be fixed to use G729 as the codec name with payload type 18.*
Linksys / Sipura SPAs do this, too, I'm pretty sure even before they were bough by cisco. *Typically this NAT functionality is referred to as a SIP NAT helper, or
SIP ALG. Of course there are other names.*
Many routers let you turn off NAT ALG, if you can find it in the menu. *Latching has several issues:... scalability* At what point does this seriously become an issue? 100 concurrent calls? 1000? If you're proxying media anyway the IP rewriting sounds pretty minimal. More info about NAT, one way media, and how sipsorcery is a SIP proxy that doesn't ever proxy media but can still set up calls with 2 way media would be much appreciated. One other things I'm interested in is the ability to not handle media, for latency & performance reasons: What are the pitfalls with FreeSWITCH specifically that makes it not happy with not handling media? and.. if I still need FS to handle DTMF what are my options? Do many/most carriers (on origination) support SIP INFO or something like that? (I run a calling card with an option to hang up the call) Thanks for the helpful info! -Avi On Fri, Jun 15, 2012 at 8:24 PM, dingdong <freeswitch@zoho.com> wrote:
** very nice and helpful,and yes,it would really be more useful if somehow related to how does FS do SIP.
Thanks
---- On Thu, 14 Jun 2012 13:06:52 -0700 *Kristian Kielhofner < kris@kriskinc.com>* wrote ----
FreeSWITCHers,
As some of you may have noticed I frequently comment on SIP-related FreeSWITCH posts. Long story short I decided to finally put some of these thoughts and experiences down in writing. What I ended up with is a 21 page (and counting) document describing various SIP header fields, SDP information, RTP issues, DTMF issues, NAT traversal technologies, interop headaches, etc.
I wrote a little intro and linked to the document if any of you would like to check it out (link at the end of post):
http://blog.krisk.org/2012/06/everything-you-wish-you-didnt-need-to.html
Of course I'd appreciate any feedback you may have. What do you disagree with? Find a typo? Have burning SIP/RTP/SDP/T.38/codec questions you'd like to see me address? Should I get into more implementation (FreeSWITCH) specific issues? Let me know.
Thanks!
-- Kristian Kielhofner
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