Outbound 3300 SIP Trunk Calls Dropping

[Wendelspanswick]

We have added some SIP trunks to a 3300 V8, inbound calls are coming in fine but outbound calls drop after about 150 seconds.
The 3300 sits on the local network behind a firewall, we have a dedicated SIP trunk gateway with a static IP of 84.252.199.200 (changed for security) on the public side of the firewall which is tunneled though to the local IP of the 3300 (192.168.1.11).
The TalkTalk SIP circuit is pointed at the endpoint which is the gateway public IP.
I'm guessing that TalkTalk are rejecting the calls as they are tagged with the local IP of the 3300 rather than the gateway IP but I can't see how to resolve this?
Or am I barking up the wrong tree.


No.     Time           Source                Destination           Protocol Length Info
      1 0.000000       84.252.199.200        91.255.255.255         SIP/SDP  888    Request: INVITE sip:07700123456@91.255.255.255 |
      2 0.001658       91.255.255.255         84.252.199.200        SIP      412    Status: 100 trying -- your call is important to us |
      3 0.838899       91.255.255.255         84.252.199.200        SIP/SDP  1013   Status: 183 Session Progress |
      4 3.468886       91.255.255.255         84.252.199.200        SIP      690    Status: 180 Ringing |
      5 6.132382       91.255.255.255         84.252.199.200        SIP/SDP  999    Status: 200 OK |
      6 6.162939       84.252.199.200        91.255.255.255         SIP      541    Request: ACK sip:07700123456@91.255.255.255:5060;alias=10.44.56.154~5060~1 |
      7 90.003357      91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
      8 90.491062      91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
      9 91.491566      91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     10 93.491550      91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     11 97.491500      91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     12 101.491455     91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     13 105.492445     91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     14 109.492394     91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     15 113.492372     91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     16 117.492313     91.255.255.255         192.168.1.11           SIP/SDP  1057   Request: INVITE sip:anonymous@192.168.1.11:5060;transport=udp, in-dialog |
     17 120.051172     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     18 120.492785     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     19 121.492257     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     20 123.493267     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     21 127.493221     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     22 131.493185     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     23 134.726536     84.252.199.200        91.255.255.255         SIP      474    Request: BYE sip:07700123456@91.255.255.255:5060;alias=10.44.56.154~5060~1 |
     24 134.728471     91.255.255.255         84.252.199.200        SIP      476    Status: 200 OK |
     25 135.493141     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     26 139.493090     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     27 143.493054     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |
     28 147.494022     91.255.255.255         192.168.1.11           SIP      581    Request: BYE sip:anonymous@192.168.1.11:5060;transport=udp |

[sunspark]

 Always dropped after 150s exactly?

[Wendelspanswick]

I've spoken to TalkTalk and it seems its the combination of the Session Timer and the local IP in the Contact Header...

The 3300 sends out a Session Refresh request for 180 seconds, TalkTalk respond after half this time (90 seconds) but its being sent to the local IP address in the Contact Header and not the public IP, the  3300 doesn't get the Refresh Confirmation so drops the call.

I can resolve this temporarily by upping the Refresh Timer or setting it to 0 (no Session Refresh request sent) but ideally I want the 3300 to send out the public IP in the Contact Header.

[sunspark]

Change the sip device capability of you sip trunk ( set to 71 )

[s4acre]

Hi

I work with "Wendelspanswick"

Can you explain in more detail please

"Change the sip device capability of you sip trunk ( set to 71 )"

There seems to be quite a few settings and none seem to fit with "71"

Thanks

[Dogbreath]

What do you mean by "SIP trunk gateway"?
- Like using an MBG to proxy it [that's the only supported way with TTB's Tipicall SIP trunks, which I assume you're using if you're connecting to  91.146.112.10]?
- Or some other model of SIP gateway [just because TTB only tested with MBG doesn't mean that others won't work]?
- Or gateway in the more general networking sense, ie a new router just for this SIP trunk?

I would leave session timer at 0 - IME, the media is renegotiated every time the session expires which doesn't help anything.

SIP peer profiles don't have SIP device capability entries so I don't know where Sunspark is referring to, maybe it's an MBG thing.

The 3300 will only ever send it's own IP in a SIP header, so if it doesn't have a public, it won't send it. If the provider expects a public IP then you need the ALG on your edge device to rewrite it or use a SIP gateway.

[sunspark]

Sorry guys. You should keep 90s for session timer and change the sip device capability for softphone users.

[x-man]

Go to your time out settings for INTERNAL call timer for outgoing call time allowed. If you see 150 secs there change that to no timer/off and your problem will go. For some strange reason sip trunks are set as internal calls on the 3300.

[Wendelspanswick]

What do you mean by "SIP trunk gateway"?
- Like using an MBG to proxy it [that's the only supported way with TTB's Tipicall SIP trunks, which I assume you're using if you're connecting to  91.146.112.10]?
- Or some other model of SIP gateway [just because TTB only tested with MBG doesn't mean that others won't work]?
- Or gateway in the more general networking sense, ie a new router just for this SIP trunk?

I would leave session timer at 0 - IME, the media is renegotiated every time the session expires which doesn't help anything.

SIP peer profiles don't have SIP device capability entries so I don't know where Sunspark is referring to, maybe it's an MBG thing.

The 3300 will only ever send it's own IP in a SIP header, so if it doesn't have a public, it won't send it. If the provider expects a public IP then you need the ALG on your edge device to rewrite it or use a SIP gateway.

The gateway is one provided by TalkTalk, its a Cisco Router managed by them and its just a data link, this connects to the customers network via the customers firewall and IT have tunneled it through their network to the IP of the 3300.
We have set the refresh timer to 9999 for now which gives us a maximum call time of 2 hrs 40 mins which is OK but not perfect.
I will have a word with IT to see if they can modify the outgoing SIP header as it passes through their firewall so that it substitutes the public IP for the internal IP.
Thank you for your help.