Hi Gavin,
I just some small extension on 1.6 branch to print more info for the
dialogs. So, right now, with dlg_list_ctx function, you can see the
profiles the dialog belongs to.
The profiles used by LB can be easily spotted by the "lbX" prefix in name.
So list all dialogs with context and see which ones are counted by LB
(based on profile) - the profile name and value will give you more info
about destination and resource.
Best regards,
Bogdan
Gavin Henry wrote:
On 6 December 2010 11:04, Bogdan-Andrei Iancu <[email protected]> wrote:
Hi Gavin,
Do you think it will be helpful for you to "see" (from OpenSIPS) which are
the these ghost calls ? (I can describe a procedure to get such a listing).
Yes, that would be great. We have tracked them down and working with
the testing vendor why they didn't send us an ACK.
Now, about what to do to prevent...More or less is about detecting so called
ghost calls....and you have several options:
1) callee/caller specific - if you are in control of one to the end points
(like you do LB over a bunch of Asterisk servers) :
We do. Our OpenSIPS LBs also do the registration and have rtpproxy on them.
A) most of the media devices (PBX, GW, etc) implements media
timeout - if no RTP is recevied, they fire a BYE, so OpenSIPS will receive
this BYE and free the call
Our media gateways (Asterisk) did do this, but OpenSIPS never got them
for some reason.
B) as A), but using signalling, SST (when the call goes via
opensips, force SST on callee side). The detection will be done by the media
server, which will send a BYE
OK, this could be good to compliment the no media.
2) opensips specific - as LB in the middle, you can check the health of
the dialog by:
A) media level - both RTPproxy and mediaproxy offers RTP timeout
events (but this approach will require media relaying for all calls)
OK, we relay media for NAT here. That will be seperate later.
B) signalling level - opensips cannot do much here at the moment
(like generating in dialog request for probing purposes)
OK.
Regards,
Bogdan
Gavin Henry wrote:
On 5 December 2010 01:07, Advantia VoIP Systems <[email protected]> wrote:
Can you verify that a BYE is sent on one UAC and and received on the
other?
We're checking. It looks like it's the technical testing calls as part
of the ITSPA Awards 2011 - http://www.itspaawards.org.uk/
We've got traces on as we speak. Even so, how can we handle not
receiving BYEs on the lb? We're open to DOS, if we get lots of these
our LBs will fill up. We have pike and ratelimit on anyway.
Thanks.
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