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<div class="moz-cite-prefix">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.<br>
<br>
I think that sofia itself supports sctp, but I don't think that
mod_sofia does. I've not looked closely to see, though.<br>
<br>
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...)<br>
<br>
-Yossi<br>
<br>
<br>
On 09/20/2013 09:07 AM, Andrew Cassidy wrote:<br>
</div>
<blockquote
cite="mid:CAMbW9oKkS2-NnNDe3zaWYe4zxkqxPBF_bnoMmHnWN2RtuSYWsw@mail.gmail.com"
type="cite">
<div dir="ltr">It's all about SCTP anyway :)</div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On 20 September 2013 14:55, Steven Ayre
<span dir="ltr"><<a moz-do-not-send="true"
href="mailto:steveayre@gmail.com" target="_blank">steveayre@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Separating SIP and RTP messages is probably
actually fairly easy since SIP messages start "SIP/2.0"
and RTP will not.
<div>
<br>
</div>
<div>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.</div>
<div><br>
</div>
<div>-Steve</div>
</div>
<div class="HOEnZb">
<div class="h5">
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On 20 September 2013 14:50,
Steven Ayre <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:steveayre@gmail.com"
target="_blank">steveayre@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">
<div>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">What
you're after is probably something like <a
moz-do-not-send="true"
href="http://tools.ietf.org/html/rfc5761"
target="_blank">http://tools.ietf.org/html/rfc5761</a> -
patches welcome ;)</blockquote>
<div><br>
</div>
</div>
<div>... that wasn't the document I was after. I
think I had seen a draft about multiplexing
SIP+RTP, but I may be mistaken.</div>
<div
style="font-family:arial,sans-serif;font-size:13px"><br>
</div>
</div>
<div>
<div>
<div class="gmail_extra"><br>
<br>
<div class="gmail_quote">On 20 September
2013 14:49, Steven Ayre <span dir="ltr"><<a
moz-do-not-send="true"
href="mailto:steveayre@gmail.com"
target="_blank">steveayre@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="margin:0 0 0 .8ex;border-left:1px
#ccc solid;padding-left:1ex">
<div dir="ltr">
<div>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span
style="font-family:arial,sans-serif;font-size:13px">Wondering why the FS
developers default to not
including rtpmap lines<br>
</span><span
style="font-family:arial,sans-serif;font-size:13px">for
statically defined RTP payload
types?</span></blockquote>
<div><br>
</div>
</div>
<div>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.</div>
<div><br>
</div>
<div>It shouldn't be such an issue
over TCP.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>
<div>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span
style="font-family:arial,sans-serif;font-size:13px">Wonder why SIP
signalling over TCP or TLS is
much more reliable</span></blockquote>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span
style="font-family:arial,sans-serif;font-size:13px">through various NAT
and firewall devices?</span></blockquote>
<div><br>
</div>
</div>
<div>That's not necessarily a given.</div>
</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>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.</div>
<div><br>
</div>
<div>
<div>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span
style="font-family:arial,sans-serif;font-size:13px">I wonder if FS could
multiplex SIP and RTP over the
same port someday?</span></blockquote>
<blockquote class="gmail_quote"
style="margin:0px 0px 0px
0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><span
style="font-family:arial,sans-serif;font-size:13px">Maybe support
deflate encoding?</span></blockquote>
<div><br>
</div>
</div>
<div>That wouldn't automatically
work. It would need support by
both ends and protocol changes to
support it.</div>
</div>
<div><br>
</div>
<div>What you're after is probably
something like <a
moz-do-not-send="true"
href="http://tools.ietf.org/html/rfc5761"
target="_blank">http://tools.ietf.org/html/rfc5761</a> -
patches welcome ;)</div>
<div><span
style="font-family:arial,sans-serif;font-size:13px"><br>
</span></div>
<div><span
style="font-family:arial,sans-serif;font-size:13px"><br>
</span></div>
<div><br>
</div>
</div>
<div>
<div>
<div class="gmail_extra">
<br>
<br>
<div class="gmail_quote">
On 20 September 2013 14:07,
Kristian Kielhofner <span
dir="ltr"><<a
moz-do-not-send="true"
href="mailto:kris@kriskinc.com"
target="_blank">kris@kriskinc.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote"
style="margin:0 0 0
.8ex;border-left:1px #ccc
solid;padding-left:1ex">
Somewhat off-topic but because
it comes up here regularly.<br>
<br>
Wondering why the FS
developers default to not
including rtpmap lines<br>
for statically defined RTP
payload types?<br>
<br>
Wonder why SIP signalling over
TCP or TLS is much more
reliable<br>
through various NAT and
firewall devices?<br>
<br>
Apple has put a significant
amount of effort into
redesigning Facetime<br>
to better handle NAT and
firewall devices. More
details here:<br>
<br>
<a moz-do-not-send="true"
href="http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html"
target="_blank">http://blog.krisk.org/2013/09/apples-new-facetime-sip-perspective.html</a><br>
<br>
I wonder if FS could multiplex
SIP and RTP over the same port
someday?<br>
Maybe support deflate
encoding?<br>
<br>
--<br>
Kristian Kielhofner<br>
<br>
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