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UDP remains the "weapon of choice" for sip trunking applications and
for many subscriber applications. However, feature-rich subscriber
implementations frequently require SIP messages to be substantially
larger than the (approximately) 1500 byte MTU (maximum transmission
unit) supported in the public internet. UDP has a mechanism for
dealing with messages greater than MTU size, but it doesn't work
reliably for SIP over UDP in the public internet. <br>
<br>
When you need to go > 1500 bytes, you need to go to TCP. <br>
<br>
Although SIP registration state and SIP call state can be replicated
over a pair of SBC's (Acme or Sonus), TCP state cannot be
maintained. There is way to much going on at the driver level for
that to be practical. When switchover occurs, the TCP "transmission
control" parameters are stale. So, there will be a TCP RST, and
the endpoints need to start over with the whole SYN / SYN,ACK
"three-way handshake" drill. <br>
<br>
Although the subscriber device will in all likelihood need to
restart the TCP session, the worst-case disruption will be only as
long as the re-registration interval. And even then, only outbound
calls (toward the subscriber device) will be affected. But even in
that unlucky instance, voice mail should save the day.<br>
<br>
The Acme "TCP session leak" was fixed several years ago At least
three. Any software having that leak has been End of Support for
long time. <br>
<br>
If anybody would like further discussion, feel free to contact me
directly.<br>
<blockquote>John S. Robinson<br>
jsr (at) communichanic dot com<br>
Communichanic Consultants, Inc.<br>
</blockquote>
<br>
<br>
<div class="moz-cite-prefix">On 7/17/2017 08:32, Pete Eisengrein
wrote:<br>
</div>
<blockquote type="cite"
cite="mid:CAHm=SaLWY9a5SqnmCoOrM7dsHdSAQsGFXxDEqyEbAr5zb+9dbQ@mail.gmail.com">
<div dir="ltr">Perfect example. With an Acme SBC as a redundant
pair, if (when) the primary fails and switches to the standby,
all UDP immediately goes to the standby and is generally
unnoticed by the end users. However, if the SIP is over TCP in
that scenario, the switch still happens but the TCP session must
re-establish itself to the secondary SBC and therefore is an
outage for those users until they re-register. I suppose it is
possible to share TCP session info, but I am not aware of any
SBC's that do this any differently (though would love to hear
from the group if one exists).
<div><br>
</div>
<div>Somewhat related, but not really: It's since been patched
but Acme (Oracle) had a bug at one point where it was not
releasing TCP sessions after they were gone and you'd end up
using all available resources (TCP ports); if that happened
you'd begin blocking new sessions and the only workaround was
a forced switchover which, of course, then meant a forced
outage for those users. </div>
<div><br>
</div>
<div>-Pete</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Mon, Jul 17, 2017 at 9:01 AM, Colton
Conor <span dir="ltr"><<a
href="mailto:colton.conor@gmail.com" target="_blank"
moz-do-not-send="true">colton.conor@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">Thanks for the replies. I am mainly talking
about using TCP for SIP signaling for <span
style="font-size:12.8px">access/customer side of the
network only. I think trunk connections to carriers will
stay UDP only for a long time. </span>
<div><br>
</div>
<div>Overwhelming it seems like using TCP for signaling
doesn't seem to be a bad thing, and preferred by many.
Peter, I have a question about what you mean by "<span
style="font-size:12.8px">But the biggest reason I
prefer UDP is for failover/redundancy. In the event of
a system failure/failover, UDP will be in-disrupted
but TCP will." </span>
<div><span style="font-size:12.8px"><br>
</span></div>
<div><span style="font-size:12.8px">Lets assume we are
using Broadsoft with an Acme Packet SBCs, and have
redundancy having one on the West coast and one on
the east cost. </span></div>
<div><span style="font-size:12.8px"><br>
</span></div>
<div><span style="font-size:12.8px">Using TCP for
signaling, how would this be different than using
UDP in a fail over secenario? Assume the client is
closer to the West cost node, and the West coast
node rebooted or shut down due to power failure. </span></div>
<div><span style="font-size:12.8px"><br>
</span></div>
<div><span style="font-size:12.8px"><br>
</span></div>
<div>Using UDP for RTP makes perfect sense. Sorry for
asking the stupid question about RTP. </div>
</div>
</div>
<div class="HOEnZb">
<div class="h5">
<div class="gmail_extra"><br>
<div class="gmail_quote">On Sun, Jul 16, 2017 at 9:59
PM, Peter E <span dir="ltr"><<a
href="mailto:peeip989@gmail.com" target="_blank"
moz-do-not-send="true">peeip989@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">The
SIP protocol already has some built in reliability
techniques built in (timers, retransmission) for
which TCP is usually used. Yes, TCP is a bit more
resource intensive due to the TCP overhead. But
the biggest reason I prefer UDP is for
failover/redundancy. In the event of a system
failure/failover, UDP will be in-disrupted but TCP
will. Your TLS argument is valid, however.<br>
<br>
RTP will always be UDP. Think about it... TCP will
retransmit when packers are lost, but in real time
communication there is no need to retransmit.
While packet loss is problematic, a retransmission
of lost packets would be unexpected and cause
further quality issues.<br>
<div>
<div class="m_-1292223908235040841h5"><br>
<br>
<br>
On Jul 16, 2017, at 22:28, Colton Conor <<a
href="mailto:colton.conor@gmail.com"
target="_blank" moz-do-not-send="true">colton.conor@gmail.com</a>>
wrote:<br>
<br>
I know UDP seems to be the gold standard for
SIP, and is in use by most service providers
that are offering hosted voice today. My
question is why not use TCP instead of UDP for
SIP signaling?<br>
<br>
Overall with small business clients we run
into firewalls with SIP ALGs, short UDP
session time out limits, and all sorts of
connectivity issues with UDP. Some small
business routers and modems have built in SIP
ALGs that can't be disabled at all. The second
we switch to TCP for signaling most of the
issues go away for our hosted voice customers.
Overall TCP just always seems to work, and UPD
depends on the situation of the network. TCP
is better for battery consumption on mobile
sip applications as well.<br>
<br>
With more providers switching to encryption
using TLS which uses TCP, is there any need
for us UDP for signaling anymore? Assuming
most IP phones from Polycom, Yealink, and
Cisco support TCP why not use it? Is it more
resouce intensive on the SBCs?<br>
<br>
What about on the media side? Does the RTP use
UDP or TCP? If it uses UDP can TCP be used?
What about for encryption like SRTP? Is SRTP
TCP or UDP?<br>
<br>
<br>
</div>
</div>
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