Avoid RTP Relay on SIP-Trunk

[derfloh]

Hi guys,

I have a problem with SIP trunk. IPO is located behind a NAT firewall. Port forwarding is not active and the provider tells me that it is not necessary. The provider doesn't have a STUN-Server.

Inbound calls work well. Outbound calls work well. My only problem is that incoming calls that are forwarded to external targets don't have audio. Signalling works well but both sides 'hear' only silence.

In SSA on SIP trunk I can see inbound an outbound call and connection type is shown as "RTP Relay". Direct Media is disabled in SIP trunk. I think if I can avoid RTP Relay all problems would be gone. RTP streams should go through IPO.

Thank you for your help.

Florian

 

[amriddle01]

Turn RTP keepalives on for that trunk

 

[derfloh]

Thank you so far. Help tells me that this is another option. But I will try that.

Sadly the customer told me to wait until tomorrow to reboot the system.

Florian

 

[tlpeter]

Florian, Are you then same one as on the trials?
You could try it with the 9.1 manager which require some less reboots.

BAZINGA!

I'm not insane, my mother had me tested!

 

[derfloh]

Hi Peter,

I am...

I tried it without a reboot (but with 9.0 Manager). But nothing changes. So I will try to reboot today in the evening.

Florian

 

[derfloh]

Hi together,

do you think it will change anything to create a new URI with another line group for outbound dialling?

I would have one URI that handles inbound call and another URI that will handle outbound calls.

Florian

 

[derfloh]

Nothing helped so far...

Another URI didn't change anything. RTP keepalives didn't change anything as well...

Do you have any other ideas?

Florian

 

[amriddle01]

RTP relay just means VCM isn't being used, the RTP does still physically pass through the IP office, you have other issues it seems

 

[derfloh]

But no idea what...

 

[intrigrant]

Try to enable "REFER" on the SIP trunk.
We have the same issue and this provider only supports external transfers with "REFER" enabled, then the transferred call is not routed through IP Office.
It has one advantage and many disadvantages :
Pro : It frees up the SIP lines
CON: IP Office loose control:
No SMDR / Chronicall data on transferred calls
No Call recording on transferred calls
No Mobile Twinning
If not answered the call doesn't return to IP Office
Maybe more but that are the issues we have.

 

[derfloh]

Hi intrigant,

thank you for that hint. I read that in documentation that REFER lets the call forward directly by the provider. But... bad luck... my provider told me that REFER is not supported. Nevertheless I tried it in any combination.

I ticked the "enable REFER" checkbox and tried to set REFER for inbound and outbound to 'never' or to 'always'. The situation didn't change.

Florian

 

[intrigrant]

Poor provider, possible one with a asterisk pc in his barn pretending to be a sip provider...

 

[derfloh]

>> Poor provider

It is a HUAWEI SBC. Don't know what type exactly...

 

[intrigrant]

Then they use either a "branding" connection, i.e. they don't have a single SIP trunk but rent them from another provider or they have not enough knowledge.
A SIP trunk is something different as a SIP extension with multiple lines.