[Asterisk-Users] Re: Asterisk SIP architecture question

[David Liu]

If you use qualify=yes to determine whether that device is alive or not, 
then it won't be very accurate as every now and then, the device may 
fail to reply to the SIP OPTIONS packet due to reasons other than it is 
really offline. 

If you are linked to a PSTN GW, I would believe that GW will monitor the 
RTP stream and then initiate a BYE if it sees no RTP packets coming in.  
That way Asterisk will receive the proper disconnect signal in a 
canreinvite=yes scenario.

David

Kevin P. Fleming wrote:

> David Thomas wrote:
>
>> Is the CDR accounting done based on SIP signaling? If a UA is talking
>> (RTP) to a third party PSTN gateway, isn't it at risk if say the UA
>> loses power. How will asterisk know the call has ended if it is not
>> involved in the media path. The idea is this.. I want to use
>> canreinvite =yes to force users to talk end-to-end to preserve
>> bandwidth, but I can see the potential for hung calls if asterisk
>> never get the BYE from a UA in the event the ATA gets unplugged or
>> somehow loses power.
>
>
> That is the case in every SIP network, Asterisk is not unique in that 
> regard.
>
> I would suggest that you could make a modification to chan_sip so that 
> if the peer goes 'unreachable' (as determined by using qualify=yes) 
> than any existing calls involved with that peer are immediately hung 
> up; that would take care of this problem.
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