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--- RE: SipGate.de & Re-Invites (http://www1.pbxes.com/forum/threadid.php?threadid=1278629752)

SipGate.de & Re-Invites

Hello,

Today, I have stumbled upon a peculiar case of an ITSP that apparently does not support SIP Re-Invites, and the inbound call gets dropped before media starts to flow, whereas outgoing calls get terminated properly without an issue.

As far as I can remember, if a Re-Invite is requested from a SIP UA, but the other SIP UA, B2BUA does not support it for any reason, the call proceeds as if the Re-Invite had never been requested.

Apparently, if the setting audio by-pass is set to Yes in the SipGate.de trunk configuration, calls using this trunk get dropped after the ringing phase, but before the media exchange phase.

Does anyone have a definitive answer whether trunks from SipGate.de support Re-Invites reliably?

I will appreciate any useful comment.

RE: SipGate.de & Re-Invites

Hi Diafora,
Not sure if it helps but,
I have set audio bypass on 2 extensions at my home and tested calls between them are ok with re-invites shown in the log.
I enabled audio bypass on my Sipgate.co.uk trunk and tested incoming calls to the same extensions. The calls complete ok but with no re-invites in the log.
I assume therefore that Sipgate does not support re-invites and pbxes has handled it correctly. Presumably Sipgate.de should behave the same way.
Regards

RE: SipGate.de & Re-Invites

Was the audio bypass set to Yes on the configuration screen of your SipGate trunk as well?

I have a SipGate.co.uk trunk as well on my account, but no SipGate.de trunk, so I haven't tested Re-Invites with it.

There is a chance the SipGate.de SIP Proxy to be different from the the SipGate.co.uk SIP Proxy, and that is precisely the reason I posted the question to the forum.

RE: SipGate.de & Re-Invites

yes,
I configured extensions and sipgate trunk with audio bypass = yes. No re-invites were seen in the pbxes log during an incoming call through Sipgate (didn't try outgoing though).
extension to extension call resulted in re-invites in the log and low latency call.
Regards

RE: SipGate.de & Re-Invites

have to check I have sipgate.de sipgate.co.uk and sipgate.com extensions.

Only use them for incoming call but never had any problems.

Is there anything I have to check for you when I receive a call on my sipgate.de number ?

RE: SipGate.de & Re-Invites

The issue was reported on inbound calls to SipGate.de DIDs, the call drops as soon as it is answered, right after the ringing phase, when the Re-Invites are activated with the "audio bypass" set to Yes on the SipGate.de trunk and the extension.

There should be System log entries like the ones below, which indicate Re-Invite activity:

VERBOSE[55702] chan_sip.c: SIP response 100 to RE-invite on outgoing call 547686622@xxx_xxx_xxx_xxx (IP of the SIP UA)
VERBOSE[55702] chan_sip.c: SIP response 200 to RE-invite on outgoing call 33b14e0443a60eaf03cd11887f30c7f1@sip.itsp.com

VERBOSE[55702] chan_sip.c: Got a SIP re-invite for call 3c2bacc3f0dc-pe6zlba00kum (Re-Invite for inbound calls)

Thanks for your help.

RE: SipGate.de & Re-Invites

Will check I am not using audio bypass but when I have some more time I will test it.