3 way calls

Sometimes, I need to have a 3 way call with my parents and A.N.Other organisation.

I was on 3 mobile, and they supported 3 way calls. My current provider, EE mobile doesn't.

I have "test" VOIP numbers with SIPgate and A&A. I have Groundwire and both accounts set-up on my IPhone.

Using Acrobits on my mobile, I can call my BT (PSTN non DV) landline, answer it, then click "add call", call the wife's mobile, the BT handset hears a "call on hold tone", my wife answers and then I "merge calls" within Acrobits for a 3 way convo.

Using the Sipgate account everything works fine.

Using the A&A account, the merge operation works according to the Acrobits screen, but the first number called, the BT landline, still hears the "on hold tone" and isn't part of the convo.

If I call the mobile number first, and add in the BT landline, the mobile continues to hear the "on hold tone" following the merge.

A&A says 3 way calling should work. I'm waiting for the wife to return with her mobile. I've enabled SIP logging within Acrobits but there is a lot of detail. I plan to retry with data disabled on my Iphone (forcing wifi only) and then wifi disabled (forcing mobile data only). I've also disconnected a Grandstream HT801 that is configured with the A&A account. I've also disabled the SIPgate account within Acrobits. Some of my actions shouldn't make any difference, but just removing them from the puzzle.

All software / IoS is up to date. I have BT FTTC with a PSTN (non VOIP) landline mainly used for incoming calls. Hopefully FTTP will be here soon.

Does anyone have A&A VOIP and Acrobits? If so, can you replicate my issue?

Thanks

More investigation has revealed that the problem isn't specific to 3 way calls. If I make a call from Acrobits to a PSTN number (PSTN to keep some separation with VOIP for debugging), I can hear myself on both devices. If I ask Acrobits to put the call on hold - for AA VOIP I get a "beep beep" call on hold tone. Unholding the call doesn't clear the "beep beep" tone.

Wireshark shows the hold/unhold. The differences between the messages (apart from fields that are meant to change - e.g. timings) is the "hold" sets "a=sendonly" in the SIP message and "un-hold" sets "a=sendrecv". RTP messages flow between both Acrobits and the VOIP server initially, and then from the VOIP server when on hold and then in both directions following the un-hold.

AA says think think it is an issue with NAT. Wireshark seems to suggest the SIP messages are working and RTP looks ok as well. AA says they have managed to get it working (by that I mean unholding and not hearing the "beep beep" tone) when 4G mobile is being used. I understood nearly all UK mobile operators used some form of NAT apart from 3 when used with a specific APN.

If SIP is unreliable when using NAT then it means Acrobits via 4G is not a solution. I guess that BT DV may use the external IP for its SIP messages to maintain reliability.

Does anyone have Acrobits and AA VOIP who can perform a hold/un-hold test? Preferably with NAT? and non NAT?

Everything works with my SIPgate number.

So no issues using your Sipgate number & Acrobits (on either LAN or 4/5G)?

@Pheasant

Thanks for the reply.

The Sipgate/Acrobits works fine on wifi or 4g/5g. Sipgate don’t play an “on hold” tone..

wireshark traces of AA and Sipgate show similar messages with no obvious differences (apart from the expected differences given different providers).

To me, it looks as though AA isn’t disabling the “beep beep”. I think hold/un-hold is not commonly used.

Sounds (pardon the pun) very much like the on-hold tone issue (“undocumented feature”) is squarely within the A&A voip platform.

I am moving towards that conclusion and have said virtually the same to AA.

Paraphrasing: SIP messages look ok for hold/un-hold and RTP flows look ok. Given SIP and RTP looks ago then how can it be NAT? Is AA processing the un-hold correctly?

That was Thursday.

I have also shared Acrobit trace logs and wire shark logs with Acrobits support. Their gut feeling is it isn’t NAT. They did say that whilst on-hold, we see SIP messages from AA which they are not expecting. They look like they are saying “AA is on hold”. Groundwire says “ ok, thanks, but I won’t do anything with this”. To me, it looks like a heart beat message which appears every 30 seconds (approx)

My IT debugging background says “is it only me that’s having issues? Can it be re-produced elsewhere? “.

Hence my request to other Acrobits/AA users

What’s the behaviour when you use another voip client other than Acrobits?

I tried Zoiper (Lite) because it is mentioned on the AA KB website. I selected the AAISP config but I could only get speech one way. The AAISP config must lock down other settings in Zoiper because I couldn't add a stun server etc.

I moved onto another AAISP listed softphone - SessionTalkSIP. This worked - I could hold/unhold. I could also get a three way call working. There was no "beep beep" on-hold tone.

I'll trace with Wireshark tomorrow and compare to Acrobits.

Strange.

In reply to a post by BarkingMad:

I tried Zoiper (Lite) because it is mentioned on the AA KB website. I selected the AAISP config but I could only get speech one way. The AAISP config must lock down other settings in Zoiper because I couldn't add a stun server etc.

I moved onto another AAISP listed softphone - SessionTalkSIP. This worked - I could hold/unhold. I could also get a three way call working. There was no "beep beep" on-hold tone.

I'll trace with Wireshark tomorrow and compare to Acrobits.

Strange.

Good that it's working correctly with AA and this SessionTalkSIP client.

As you say this isn't an oft used feature, and I suspect there is a subtle difference in the SIP messaging / implementation / settings which is tripping up between AA and Acrobits. NAT to me is a total red herring in this situation, call flows, RTP and signalling seems fine. Its at a deeper level and not network.

I've only experience using Acrobits (GroundWire) and Sipgate, so can't offer any help with the AA service, sorry.

Hi Pheasant,

I looked at Wireshark debug for the SessionTalkSIP. It was similar to Acrobits and no obvious differences. I've sent the trace to AA so they can compare although I get the view VOIP (with NAT) isn't top of their list when it comes to product support.

I have noticed that within Acrobits there is an AAISP account profile. I've tried that and it doesn't work. The The AAISP help pages https://support.aa.net.uk/VoIP_Phones_-_Acrobits do contain some setup info for Acrobits but it looks like it is "work in progress".

I have a solution for 3 way calls which should meet my occasional need for the functionality.

I think it is something on the AAISP side this isn't working correctly for whatever reason.

Thanks for your help.

I'm hoping to have FTTP with a fixed IP early next year (fingers crossed)

In reply to a post by BarkingMad:

I have a solution for 3 way calls which should meet my occasional need for the functionality.

I think that's as good as you can hope for / expect. I don't think VOIP is particularly remunerative for AA (or anyone really in this particular part of the market) - probably therefore not very high on their list of priorities. But you never know.
👍

Edited by Pheasant (Thu 24-Oct-24 13:32:58)

I needed to use a 3 way call to speak with my parents and insurance provider.

Using SessionTalk and AA, which had been proven to work for 3 way calls, I called the insurance provider, went through the menu system, 1,2,2,1 and then got a continuous tone instead of speaking to the correct department. Repeated a few times. Used my EE mobile and OK to speak to someone.

I worked around with a 3 PAYG SIM in another phone.

I've just tested with Acrobits/AA and same issue. However, Acrobits/SIPgate was fine.

I haven't done a wireshark trace or informed AA because I'll just be informed that "NAT is evil". However, SIPgate is fine with NAT.

I suppose I could look at allowing my non-static BT FTTC IP (which I understand could change) through the firewall on a non BT router - to see if I have the same issues with AA.

I'll need to google to find a suitable guide so I don't end up with a security issue!

Yeah it’s not NAT (or the client) given this all works with Sipgate and Acrobits (same combo I’m using).

Points to something within the AA VoIP platform / specific to their implementation.

I don’t think messing around with your firewall settings or looking at traces from Wireshark will bear much fruit.

I reckon it’s squarely in AA’s court. Just a corner / oddball case.

Edited by Pheasant (Thu 07-Nov-24 17:53:07)

Thanks Pheasant,

It looks like I'll need to keep my basic/free SIPgate "test" VoIP number for the foreseeable future because it is the only VoIP (with NAT) which can perform 3 way calls and connect to an insurance department following an interactive menu.

I'll probably use AA to port my BT PSTN when FTTP arrives.

I may look around for a low cost VoIP provider just in case SIPgate pull the free product.

I had planned to use the VoIP when roaming (with a data E-SIM), plus the the occasional 3 way call, and just to keep the PSTN number just in case someone didn't have our mobile numbers.