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<p class=MsoNormal><span style='color:#1F497D'>Sounds like H323 is your
protocol then. </span><span style='font-family:Wingdings;color:#1F497D'>J</span><span
style='color:#1F497D'><o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>SCCP will require 1-1.5 x <RTT
latency> to setup a full audio stream. CUCM will send the device an
OpenLogicalChannel, the device will process it and respond with an
OpenLogicalChannelAck, and then CUCM processes the Ack and generates a
StartMediaTransmission to send to the other party (in the case of two SCCP
phones).<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>So if you have a 600ms RTT from
site A to siteB, then you have to wait for the OLC to make it there (~300ms),
then wait for the Ack to be returned (~300ms), then CUCM has to send the SMT to
the other party (time here depends on where this other device is located and
whether or not it is going to experience any delay in transit); hence 1-1.5 x <RTT
latency>.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'>So in your case, using SCCP, you
are looking at roughly 1+ seconds after off-hook to get full two-way audio.
There is not “faststart” SCCP that I am aware of.<o:p></o:p></span></p>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<p class=MsoNormal><span style='color:#1F497D'>-Chris<o:p></o:p></span></p>
</div>
<p class=MsoNormal><span style='color:#1F497D'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> cisco-voip-bounces@puck.nether.net
[mailto:cisco-voip-bounces@puck.nether.net] <b>On Behalf Of </b>Keith Klevenski<br>
<b>Sent:</b> Monday, August 03, 2009 5:19 PM<br>
<b>To:</b> Wes Sisk (wsisk)<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Speed up skinny?<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'>Wes,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'>Thanks for the reply. Running CM 7.02. I’m
quite sure its RTT as we’re talking 600-800ms with 100 byte packets
although the link speeds are 128/256kbps so we’re not breaking any speed
records here. ;) If I change the SCCP gateway to H323 and configure
the usual dial-peers the delay is gone. Convert back to SCCP and
it’s back. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'>I don’t see any way around it based on the round trip
requirement you mentioned. Thanks for the confirmation. <o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'><o:p> </o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'>-Keith<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif";
color:windowtext'><o:p> </o:p></span></p>
<div>
<div style='border:none;border-top:solid #B5C4DF 1.0pt;padding:3.0pt 0in 0in 0in'>
<p class=MsoNormal><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif";
color:windowtext'>From:</span></b><span style='font-size:10.0pt;font-family:
"Tahoma","sans-serif";color:windowtext'> Wes Sisk [mailto:wsisk@cisco.com] <br>
<b>Sent:</b> Monday, August 03, 2009 3:39 PM<br>
<b>To:</b> Keith Klevenski<br>
<b>Cc:</b> cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] Speed up skinny?<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'>What CM version are you on?<br>
<br>
Only recent optimization available is Advanced CM service parameter
"Bundle Outbound SCCP Messages Timer" to enable bundling of SCCP
messages. More messages per packet = fewer ACKs = fewer round trips. Much
further back there was an effort to streamline the number of messages required
to setup SCCP call.<br>
<br>
SCCP capabilities are exchanged at device registration so it does not suffer
the same issue as h.323 where capabilities must exchange during every call
setup. There is no MSD equivalent. SCCP overall is a much lighter
protocol at the point of call signaling. The only round trip requirement
in SCCP is OpenReceiveChannel/OpenReceiveChannelAck with remote IP:Port.
StartMediaTransmission is required and cannot be sent until IP:Port is received
from far end device. Unfortunately this exchange is required with no way
to bypass this round trip for SCCP.<br>
<br>
One area we commonly see issues is shared lines at remote sites. This is
covered in SRND:<br>
<a
href="http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/netstruc.html#wp1045582">http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/srnd/6x/netstruc.html#wp1045582</a><br>
<br>
Is it really RTT or is it possibly link speed that is causing the issue?<br>
<br>
/Wes<br>
<br>
<br>
<br>
On Monday, August 03, 2009 4:11:50 PM , Keith Klevenski <a
href="mailto:KKlevenski@cstcorp.net"><KKlevenski@cstcorp.net></a> wrote:<o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Is
there any mechanism to ‘speed up’ skinny call setup like faststart
in h323? Satellite provider migrated to skinny gateways (SCCP controlled
FXS ports) from h323 gateways. All analog phones. It was noticed
that inbound PSTN calls (through h323 gateway) were losing the first second or
so of the call when the called party picked up. This made sense to me as
the skinny gateway must tell CM the phone has gone offhook then CM tells the
gateway to tell the phone to stop ringing and send media stream to the PSTN
gateway. This is over a 600-700ms satellite link so the delay would be
reasonable. The rest of the call is perfect. If the same skinny
gateway is changed to h323 the delay is not present since h323 gets the IP
address where to send the stream during call setup if I’m not mistaken so
the delay isn’t present, but neither are the features of skinny
controlled fxs ports…</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>To
me this is all reasonable and makes perfect sense. The question the
customer has is if there is any mechanism to speed up the skinny
signaling. I wouldn’t think so based on the master/slave nature of
the skinny protocol. I’ve configured voice call send-alert, voice
rtp send-recv, progress_ind setup enable 3, and all that, but nothing helps and
I can understand why.</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Can
anyone confirm if there is a way to speed up skinny call setup over high
latency links? Running CM 7.0, SCCP controlled FXS.</span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'> </span><o:p></o:p></p>
<p class=MsoNormal><span style='font-size:10.0pt;font-family:"Arial","sans-serif"'>Thanks!<br>
<br>
-Keith </span><o:p></o:p></p>
<pre><o:p> </o:p></pre><pre style='text-align:center'><o:p> </o:p></pre><pre
style='text-align:center'><o:p> </o:p></pre><pre style='text-align:center'>
<hr size=4 width="90%" align=center>
</pre><pre style='text-align:center'><o:p> </o:p></pre><pre
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