[Freeswitch-users] OT: Apple goes to great lengths to defeat NAT and firewalls with new Facetime

I put the Who? in Mishehu mishehu at freeswitch.org
Sat Sep 28 03:42:19 MSD 2013


Well SCTP operates at the same level as UDP and TCP.  Unfortunately not 
enough systems support it, though I have done some work with SCTP in the 
lab in the past - even for media streams, not just sigtran.  I never did 
the "send packets at the server until smoke comes out" type of load 
testing, so I don't know if the added complexity in sctp would be cause 
for lower capacity per server.  However, another upshot is that it 
(sctp) can even mux multiple streams into a packet.

I think that sofia itself supports sctp, but I don't think that 
mod_sofia does.  I've not looked closely to see, though.

I for one would welcome our new SCTP overlords if and when they ever get 
here.  (I'm still waiting for our ipv6 overlords to arrive as well...)

-Yossi


On 09/20/2013 09:07 AM, Andrew Cassidy wrote:
> It's all about SCTP anyway :)
>
>
> On 20 September 2013 14:55, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     Separating SIP and RTP messages is probably actually fairly easy
>     since SIP messages start "SIP/2.0" and RTP will not.
>
>     Architecturally the SIP and RTP stacks are implemented by separate
>     libraries though, and having them both bound to the same port
>     would be far from trivial (both libraries would need to be
>     rewritten to go via some abstraction layer). It'd also only be
>     possible to do it for UDP.
>
>     -Steve
>
>
>     On 20 September 2013 14:50, Steven Ayre <steveayre at gmail.com
>     <mailto:steveayre at gmail.com>> wrote:
>
>             What you're after is probably something like
>             http://tools.ietf.org/html/rfc5761 - patches welcome ;)
>
>
>         ... that wasn't the document I was after. I think I had seen a
>         draft about multiplexing SIP+RTP, but I may be mistaken.
>
>
>
>         On 20 September 2013 14:49, Steven Ayre <steveayre at gmail.com
>         <mailto:steveayre at gmail.com>> wrote:
>
>                 Wondering why the FS developers default to not
>                 including rtpmap lines
>                 for statically defined RTP payload types?
>
>
>             Because SIP over UDP doesn't play well with fragmentation.
>             SDP can make packets large than the PMTU, leading to
>             fragmented packets, which leads to devices ignoring the
>             packet. Removing the unnecessary rtpmap lines means
>             smaller SDP so smaller packet so less likelihood of that
>             being an issue.
>
>             It shouldn't be such an issue over TCP.
>
>             Devices *should* support it since the standard explicitly
>             say the rtpmap isn't required for static types, but there
>             are some manufacturers who ignored that part so there's
>             the verbose_sdp=true compatibility option for them.
>
>                 Wonder why SIP signalling over TCP or TLS is much more
>                 reliable
>
>                 through various NAT and firewall devices?
>
>
>             That's not necessarily a given.
>
>             In general though because the TCP connection explicitly
>             signals the connection closing the mapping will stay in
>             the firewall. With UDP it is removed after a long period
>             of inactivity. That can cause problems with signalling
>             during a phone call unless the endpoints send keepalive
>             packets often enough.
>
>             TLS will prevent the router helping with SIP ALG - you
>             must have endpoints capable of doing NAT traversal
>             themselves (STUN). Though that's a good idea in all cases
>             anyway.
>
>                 I wonder if FS could multiplex SIP and RTP over the
>                 same port someday?
>
>                 Maybe support deflate encoding?
>
>
>             That wouldn't automatically work. It would need support by
>             both ends and protocol changes to support it.
>
>             What you're after is probably something like
>             http://tools.ietf.org/html/rfc5761 - patches welcome ;)
>
>
>
>
>
>             On 20 September 2013 14:07, Kristian Kielhofner
>             <kris at kriskinc.com <mailto:kris at kriskinc.com>> wrote:
>
>                 Somewhat off-topic but because it comes up here regularly.
>
>                 Wondering why the FS developers default to not
>                 including rtpmap lines
>                 for statically defined RTP payload types?
>
>                 Wonder why SIP signalling over TCP or TLS is much more
>                 reliable
>                 through various NAT and firewall devices?
>
>                 Apple has put a significant amount of effort into
>                 redesigning Facetime
>                 to better handle NAT and firewall devices.  More
>                 details here:
>
>                 http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html
>
>                 I wonder if FS could multiplex SIP and RTP over the
>                 same port someday?
>                  Maybe support deflate encoding?
>
>                 --
>                 Kristian Kielhofner
>
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