[Freeswitch-users] Blog post about SIP

[Michael Giagnocavo]

Excellent work. It's great to see a practical guide to the complicated (and sometimes downright moronic) mess that encumbers VoIP.

Some notes:

- From viewing a medium sized production network, I see about 1% of the SIP packets received and processed are fragmented (MTU of 576 bytes somewhere along the way). Are you seeing implementations actually failing due to IP fragmentation of UDP packets? I tested sending 2KB length UDP messages to my proxies over the Internet, and it seems to work. Certainly sending frag'd SIP messages is more likely to work than the crazy idea of protocol transition, right? Who's really experiencing MTU issues?

- Watch out with ICE and VPNs. While communicating to a server on a public IP that's also available internally via VPN, an aggressive anti-NAT solution like ICE might determine there's a "direct" path via the private VPN IPs. If the VPN is not a good path (say, its implemented over SSL), connectivity will happen over that path, and audio quality suffers. Microsoft Lync does this (Google lots of complaints related to using Lync with a VPN), and the only workaround is to configure firewalls to block such traffic.
	- Subnote: I wonder how well things would work if SIP clients implemented UPnP, which is supported by tons of home routers?

- Re-INVITE messages *should* be distinguished by a to tag. But, there are many networks out there using software written by folks that didn't bother to read the SIP spec, and they will happily ignore an existing to tag and treat it like a new request. This has implications for the standard OpenSIPS config, for instance, which allows relaying of INVITEs if there's a to-tag.

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Kristian Kielhofner
Sent: Thursday, June 14, 2012 2:07 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Blog post about SIP

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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