IPO R11 + DECT Siptrunk problem UPDATE - different scenario's

[somer10]

Hi All,

A bit a last resort question here as Avaya does not want to help without a DECT support contract and the provider is pointing at the pbx as the issue.

We have a customer with multiple sites, same setup everywhere.
IP Office R11 + DECT installation + siptrunk of Proximus (Belgium)

Only on 1 site we have a problem with DECT phones calling to mobile phones.

50% of the calls are OK
25% of callsetups are timedout after 30 seconds.
25% of calls are OK but only after 30 seconds.

IP phones,digital phones do not have the problem.

We placed a test pbx +dect installation at site A where the problem occurs. Result = Also our test pbx has this problem
We placed the same test pbx+dect installation at site B. Result= the problem does not occur there.

-> this rules out that the pbx/dect is the problem because same pbx works on site B but no on site A.
-> looking at this test it would suggest that this is a provider problem at site A.

But how do we explain why this issue is not happening when using an ip or digital phone on this location.
-> this suggests that the dect installation is the problem.

Avaya wants a dect support contract to investigate, Provider says problem is the pbx.

[somer10]

Analysis: all calls were picked up immediately by the mobile phone so no delayed media should be present.All calls were made from same user to the same mobile phone
When a DECT phone calls out to a mobile phone we have 3 possible scenarios:

1. Scenario1: Call is good
2. Scenario2: Call is bad:
   dect phone dials out -> mobile phone immediately takes the call. DECT phone keeps hearing ringback tone for +30seconds even though the mobile phone has already taken the call. The mobile phone does not hear anything. After 30 seconds the media path is established and we have 2way audio.
3. Scenario3:Call is bad:
   same scenario but after 30 seconds we receive a 408 request timeout from the provider and the call is dropped.

Below are the call scenarios from left to right (scenario 1-2-3)

• S1: good call: pbx receives 183, pbx send media after it.
• S2: bad call: pbx receives 183,pbx does not send media after the 183. pbx receives an update from provider but pbx does not respond to it. After 30 seconds the pbx receives an 200OK answer on the initial invite and the media is being send in both directions
• S3: bad call: pbx receives 183,pbx does not send media after the 183. pbx receives an update from provider but pbx does not respond to it. After 30 seconds the pbx receives 408 request terminated from the provider.

[somer10]

So 3 question immerge here: bearing in mind that these scenarios do not occur on the same test pbx at a different location:

Why does the pbx not respond by sending RTP stream on the 183 session in progress?

In call 1 the pbx sends RTP

In call 2 and 3 it does not. Same destination/source as call 1

subquestion: Is it mandatory that the pbx responds to the 183 by sending rtp like in scenario1?

Why does the pbx not respond on the UPDATE SDP messages in scenario 2 and 3?
subquestion: Must the pbx respond to the UPDATE message from the provider before the provider forwards the 200OK respons from the initial invite?

The provider told us they hold back the 200OK response on the initial invite because the pbx did not respond to the UPDATE message. The update message does not contain any additional info compared to the 183 that was already received. The update is actually a converted 'ringing' which the provider converts for some reason.

Why does the provider respond differently on scenario 2 and 3 while the flow is exactly the same?