[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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> FreeSWITCH-powered IP PBX: The CudaTel Communication
> Server
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <mailto:info at cassidywebservices.co.uk> *03300 100 960 *F
> <mailto:info at cassidywebservices.co.uk> *03300 100 961
> *E <mailto:info at cassidywebservices.co.uk>
> *andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
> *W <mailto:info at cassidywebservices.co.uk>
> *www.cassidywebservices.co.uk <http://www.cassidywebservices.co.uk>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
>
>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20130927/832b241a/attachment-0001.html
Join us at ClueCon 2013 Aug 6-8, 2013
More information about the FreeSWITCH-users
mailing list