asterisk users - Feb 2005 - SIP Qualify/Status – What kind of numbers are you getting?

I have several Polycom IP-500?s and a few of the Cisco 7960?s connected 
to an Asterisk test box.   When I add qualify=yes to the sip.conf and 
then enter ?sip show peers? on the console I get, on average, 85 ms for 
the Polycom phones while the Cisco phones are half that.  This is on a 
LAN.   Across a T1 the Polycom phones are general around 100 ms or more.

What kind of status value is common for what kind of phone?
Is the Polycom just slow to reply?

I did not have any calls active in the system while gathering this 
information and all phones are on real addresses (no nat).   I also 
verified that the timing was correct with Ethereal.   Ping tests from 
the * box to all the phones in question were <1 ms. I know that overhead 
for ICMP is nothing compared to a UDP/SIP packet but the delta is more 
than expected.   Does the ?SIP OPTIONS? create that much overhead?

More info from the WIKI:
<snip>  
SIP.conf: device configuration - qualify
Syntax:     qualify=xxx|no|yes
where XXX is the number of milliseconds used. If yes the default timeout 
is used, 2 seconds.

If you turn on qualify in the configuration of a SIP device in sip.conf, 
Asterisk will send a SIP OPTIONS command regularly to check that the 
device is still online. If the device does not answer within the 
configured (or default) period (in ms) Asterisk considers the device 
off-line for future calls.

This feature may also be used to keep a UDP session open to a device 
that is located behind a network address translator (NAT). By sending 
the OPTIONS request, the UDP port binding in the NAT (on the outside 
address of the NAT/firewall device) is maintained by sending traffic 
through it. If the binding were to expire, there would be no way for 
Asterisk to initiate a call to the SIP device. This can be used in 
conjunction with the nat=yes setting.
</snip>

Thanks
Scott H

We have a mix of Polycom IP600 and Sipura SPA2000 devices across our 
network. I have noticed that the response times for the Polycom are 
significantly higher than for other devices. I have also tested Cisco 
and Snom hard phones. All of our phones are on T1 links back to the * 
server.

My times for the Polycom in an idle state are generally 70-85ms while 
the Sipura devices run <20ms. the Polycoms seem to be around 120ms when 
inuse.

Hope this helps.

On Feb 8, 2005, at 4:52 PM, Scott Herrick wrote:
> I have several Polycom IP-500?s and a few of the Cisco 7960?s 
> connected to an Asterisk test box.   When I add qualify=yes to the 
> sip.conf and then enter ?sip show peers? on the console I get, on 
> average, 85 ms for the Polycom phones while the Cisco phones are half 
> that.  This is on a LAN.   Across a T1 the Polycom phones are general 
> around 100 ms or more.
>
> What kind of status value is common for what kind of phone?
> Is the Polycom just slow to reply?
>
> I did not have any calls active in the system while gathering this 
> information and all phones are on real addresses (no nat).   I also 
> verified that the timing was correct with Ethereal.   Ping tests from 
> the * box to all the phones in question were <1 ms. I know that 
> overhead for ICMP is nothing compared to a UDP/SIP packet but the 
> delta is more than expected.   Does the ?SIP OPTIONS? create that much 
> overhead?
>
> More info from the WIKI:
> <snip>  SIP.conf: device configuration - qualify
> Syntax:     qualify=xxx|no|yes
> where XXX is the number of milliseconds used. If yes the default 
> timeout is used, 2 seconds.
>
> If you turn on qualify in the configuration of a SIP device in 
> sip.conf, Asterisk will send a SIP OPTIONS command regularly to check 
> that the device is still online. If the device does not answer within 
> the configured (or default) period (in ms) Asterisk considers the 
> device off-line for future calls.
>
> This feature may also be used to keep a UDP session open to a device 
> that is located behind a network address translator (NAT). By sending 
> the OPTIONS request, the UDP port binding in the NAT (on the outside 
> address of the NAT/firewall device) is maintained by sending traffic 
> through it. If the binding were to expire, there would be no way for 
> Asterisk to initiate a call to the SIP device. This can be used in 
> conjunction with the nat=yes setting.
> </snip>
>
> Thanks
> Scott H
>
>
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--
Jerry Jones
(763) 201-1266