Call Forward to external received CANCEL after ringing

[Jonathan Galvez]

Hi folks,

I have a problem with my SIP-Provider. When I make a external call forward they send me a CANCEL after that Phone rings one time.

SIP-Provider => CUBE => CUCM (call forward) => CUBE => SIP-Provider

Incoming, Outgoing and transfer calls work. 

The provider tell me that we need to send a few empty RTP packets or have "comfort noise on" permanently so that the line is not "dead". Do you know if that is possible on a CISCO CUBE? I have tried with STUN but it didn't help...

Many thanks, Jonathan

[b.winter]

No, CUBE cannot send dummy RTP packets.
Only CUCM can do that, if you enable MTP on the SIP Trunk towards CUBE. (but MTP should be avoided in general, as it produces higher load on CUCM)

Edit:
What you also can try is to configure media anti-trombone

voice service voip
 media anti-trombone

[Jonathan Galvez]

unfortunately doesn't work with MTP, still receive the CANCEL from provider. Should I deactivate STUN to configure the anti-trombone?

[b.winter]

Then you might have a different problem.

Could you post the config (without any sensitive data) and the output of the following debugs for a test call?
debug voice ccapi ind 1
debug voice ccapi ind 2
debug voice ccapi ind 74
debug ccsip messages

[b.winter]

But problems with your scenario are common with SIP providers I have worked with and many of these problems have been discussed already in the forum.
E.g. here: https://community.cisco.com/t5/unified-communications-infrastructure/cube-config-sip-trunk-to-deutsche-telekom/m-p/4455797#U4455797

But as already written, I don't think that the Cancel is caused by the missing RTP packets.
Normally you would have a call establishing, but with no audio in both directions.

Maybe the call is not allowed, because normally SIP providers often use the FROM, PAI or PPI header for authentication, and if you have in there a number not assigned to your DID range, then the provider will decline the call.

And in the scenario PSTN --> CUCM --> PSTN, CUBE will send the external calling number in the FROM (normal behavior), which is obviously not a number of your DID range, then the provider will decline it.
As written, most providers then look at the PAI or PPI header, if there is a number of your DID range (which you could configure in the CUBE with SIP profiles), but you have to ask the provider such question or you have a doc from them.

[Jonathan Galvez]

I hope, it is clear for you.

They just said, that they need this empty rtp...I am confused because if they do not like my INVITE why the phone rings one time? That is weird...I will parallely have a look at the post, thanks!

[b.winter]

 

I don't think, that the answer you got from the provider is accurate. Because the call isn't even established, so RTP is not even in place.

Just a guess:
Maybe the provider cancels the call, because he didn't receive a 180 Ringing for a certain time period.
Maybe he doesn't like the 183 Session Progress and/or ignores it.
Since there is no 180 Ringing or 200 Ok after x seconds, the provider cancels the call.

As you can see in the screenshot, CUBE converts the 180 Ringing with SDP (received from provider) to a 183 Session Progress with SDP (sent to CUCM).
And CUBE also gets 183 from CUCM, because CUCM just uses the same method.

And if CUBE receives a 183 on one leg, it also sends a 183 on the other leg.

Long story short:
Try to configure "send 180 sdp" in the dial-peers or globally in "voice service voip"

 

[Jonathan Galvez]

so I am done for today but I did the following:

deactivated 180 messages with "disable-early-media 180"

Now I can stablish the call but still without Audio...

[b.winter]

Then now you have the problem, which I described in the other post.
Try with MTP.

[Jonathan Galvez]

Yeah it works but I can not keep this active, I need another solution

[b.winter]

But you only have the options described in the other post:

• Public IP on the CUBE (best and least problem making solution)
• STUN config
• media anti-trombone
• MTP on SIP Trunk (last resort, always working with that, but the worst solution)

There are no other options.

[b.winter]

@Jonathan Galvez Your stun config cannot work, because you haven't assigned the stun-usage to no dial-peer.
Config voice-class stun-usage 1 to every dial-peer and try if you have audio then (without MTP)

[Jonathan Galvez]

Hi, I tried and didn't work. You are right on the config I sent to you was not applied, cause I change so many things and you got the wrong config

[Jonathan Galvez · Accepted solution]

Hello,

just to close the post :), I got the solution, is there another possibility without MTP to solve the audio problems when STUN or media Anti-Trombone doesn't work. 
voice service voip
sip
nat force-on

Hier a link how it works:
https://www.cisco.com/en/US/docs/routers/asr1000/configuration/guide/sbcu/sbc_nat.pdf

That solve my issue, thanks an Ahmed from TAC!