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

[Scott Herrick]

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