Re: [Asterisk-Users] SIP Qualify/Status – What kind of numbers are you getting?

[jjones at quiddesign.com]

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