ptime=20 vs ptime=60 issues

[InsaneSplash]

Hello,

When 3CX recieves an inbound call with a ptime=20, the system routes the call correctly, ie to an IVR or extension. If the ptime received is not ptime=20, ie ptime=60, the external caller continues to hear the call ring, however 3CX "picks" up the call and begins its routing process. The caller eventually just hangs up.

We have tested using multiple inbound sources, and each time the call shows a ptime that is NOT 20, the call issue presents itself. Is there a bug with 3CX that it does not support any ptime's higher than 20?

Working Trace:

10/11/2023 10:53:41 - [CM500002]: Call(C:404): Info on incoming INVITE from Line:10000<<++27000000000:
Invite-IN Recv Req INVITE from a.b.c.d:5060 tid=7d47f874 [email protected]:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK7d47f874;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "0000000000" <sip:[email protected]>;tag=as23f5f0c8
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Date: Fri, 10 Nov 2023 08:53:41 GMT
In-Reply-To: SDilqsd01-627159de930d0426b675cdf68d3d575e-ctvvfv3
Supported: replaces, timer
User-Agent: Telviva N+
Diversion: <sip:111111111@jt-as08>
Content-Length: 350

v=0
o=root 1466536994 1466536994 IN IP4 a.b.c.d
s=Telviva
c=IN IP4 a.b.c.d
t=0 0
m=audio 13592 RTP/AVP 107 9 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv

Failed Trace:

10/11/2023 10:53:31 - [CM500002]: Call(C:403): Info on incoming INVITE from Line:10000<<++27000000000:
Invite-IN Recv Req INVITE from a.b.c.d:5060 tid=05e2cd4c [email protected]:5060:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP a.b.c.d:5060;branch=z9hG4bK05e2cd4c;rport=5060
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "000000000" <sip:[email protected]>;tag=as63720dce
Call-ID: [email protected]:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Date: Fri, 10 Nov 2023 08:53:31 GMT
In-Reply-To: SDejftd01-cceddb47743b809736b9ad70ee99ca11-ctvvfv3
Supported: replaces, timer
User-Agent: Telviva N+
Diversion: <sip:111111111@jt-as06>
Content-Length: 348

v=0
o=root 642661811 642661811 IN IP4 a.b.c.d
s=Telviva
c=IN IP4 a.b.c.d
t=0 0
m=audio 15788 RTP/AVP 107 9 8 0 101
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:60
a=maxptime:60
a=sendrecv

 

[Alejandro_3CX]

Hi @InsaneSplash I'm afraid that Telviva N+ is not listed as supported voip provider. From 3cx we encourage all the partners and users to use recommended voip providers in this list. These has been tested and approved the interoperability test so we can guarantee the correct integration with 3CX and know the limitations that may exists with each sip trunk.

Have a great weekend!

 

[InsaneSplash]

Hi @Alejandro_3CX, Thanks for the reply. As there are only 2 voip providers in your list for South Africa, this does limit the use of the product within the country. There are far more such as carrier telcommunications companies (national carriers) that offer SIP trunks. As we know, voip providers use a number of backend systems, such as Asterisk, PortaOne etc to provide a SIP trunk service and one can't really say that the above issue has anything to do with this. The affected calls are linked to local mobile network providers.

The reply doesn't really answer the question around the ptime issue as this is a RFC standard which 3CX seems not to be honouring. Why would 3CX not accept a ptime thats offered within the maxptime it reports to support?

How does a provider join the recommended voip providers list?
What tests need to be performed to pass the interoperability test?

 

[jed]

What tests need to be performed to pass the interoperability test?

For the Interop, have a look here: https://www.3cx.com/docs/supported-sip-trunk-requirements/

 

[InsaneSplash]

@jed Thanks for the reply, I see in the linked docs there are no specifics around for example ptime. I do however see that the SIP trunk we are using is "compliant" based on the requirements that are mandatory and optional except for SMS support.