Apologize for the late response. As per the higher tier of NETGEAR Support, while Auto VoIP alone will provide the segregation and the prioritization, it is best practice to apply the Voice VLAN setting also to tell the phone which VLAN to tag itself to, this could be important when there is a phone connected to a switch port and there is a PC connected into the phone.
for a network like ours where phone can be part of different vlans [ i set up using "vlan static routing wizard] - auto-voip seems to be all that is needed to make sure voip traffic get high priority.
[ another example is lag set up - as i posted in another post it is very different to set up - similiar in some ways but a wire must be attached to change lag type. so documentation or examples for other switches has missing important data].
We have similar traffic shaping rule setup on MX64 to handle voip traffic. We migrated to hosted voip about 4 weeks ago adding that traffic to both existing sites using MX64 appliances. We've encountered problems, at both of our sites, since migration. The Avaya phones intermittently re-register and retrieve their settings...If this event occurs while phone in conversation all ability to manage the call is gone... (conversation continues however ability to manage call is lost. Cannot transfer, conf, forward, hang-up etc....) Our phone vendor believes the problem to be result of poor jittery internet connection or MX configuration which is not optimum for VOIP traffic. I'm still in the process of eliminating what IS NOT contributing to this behavior while pushing on noise found on internet paths.
Anyone have input into best configuration to shape voip traffic on MX64? To avoid the behavior in one of the buildings we're moving VOIP Traffic in/out of building over different internet path. This means we've also isolated Data/Voip. This solved issue in one building and results would appear to indicate the problem to be exclusively quality of internet connection, however can't rule out the separation of Data and Voice as the real solution. (Noting that Data is remains uninterrupted on same internet path were VOIP was encountering all kinds of trouble.) Of course VOIP traffic less forgiving to jitters or poor internet connections. One distinction between the traffic shaping screen shot posted and our rule are the limits on bandwidth for VOIP traffic set to 5Mbps rather than unlimited. This would seem most important, however other building with same behavior had no traffic shaping rule and still encountered same problem phones reregistering and retrieving their settings.
Then you need to measure you upstream bandwidth (using a tool like www.nperf.com or www.speedtest.com) and then configure this number as your WAN uplink bandwidth (so the MX actually knows how much bandwidth you have available).
I would also upgrade to 14.31 firmware. Then setup a traffic shaping rule and change VOIP traffic to DSCP 46. They changed the functionality of traffic shaping in 14.x fimrware with traffic that has DSCP 46.
Are you using Meraki MS switches? I would recommend go got "Switch > Switch Settings" and setting the DSCP value for your phone VLAN to 46. Then in traffic shaping on the MX have it NOT change the DSCP value, it can still prioritize VoIP traffic though. This will set the traffic priority higher closer to the VoIP phone so as it traverses your internal network it has the higher priority, then when it gets to your MX you don't want it to be slowing that traffic down by changing the DSCP value again.
I think your issue sounds like it could be related to UDP timeout which you cannot change on the Meraki. You could try changing the timeout on your Avaya system to be longer though. Too much jitter would more likely cause poor audio quality, but not phones re-registering like you describe.
Because of the latest critical vuln, I needed to upgrade all 50 of my 80/90E fortigates to 7.0.9 from 6.4.x ( I could have gone 6.4.11 but decided it was a good time to make the leap to 7.0.x). Now, I was hasty and didn't fully test, I admit it so don't worry I yelled at myself. Here's what I ran into....these FGT's host Mediatrix FXS devices C700. Basically its a device that will SIP register a certain number of lines to our PBX and then hand that voip signal off via analog. We use a static VIP on the FGT to do a translation and all of the mediatrix are hard-coded with the same local IP; and the FGT translates it outbound (this design was in place before I took over the network, so no comment here).
Basically, I went from 6.4.x (we had many flavors); up to 7.0.9 using Fortimanager to use the approved upgrade path on all. and as soon as I did that it broke SIP registeration on every single one, because the PBX started seeing the internal IP in the register packet instead of the external. I did pcaps on the internal and external interface both before and after, and confirmed that despite the config not changing at all (which support validated), the fgt is handling the packets differently.
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So with that said: I don't know for sure. I believe that's correct, but can't verify it from experience. I don't know what a "VOIP services certificate" is for, but I'm confident that "normal" APNs certificate can send notifications to com.example.yourapp, com.example.yourapp.voip, and com.example.yourapp.complication. There may be more that needs to happen in the wider VOIP ecosystem, but in terms of sending an APNs notification to a VOIP topic, the non-VOIP certificate should be just fine.
If a device turn off an app's notification (Setting -> app -> notification -> off), the app can't receive push notification from push kit => Unable to receive VoIP push => unable to handle VoIP call if app's inactive. I tested this case but still hope someone else can confirm this.
However, there are some other apps (e.g. facebook messenger), even with disabled notification setting, can still receive VoIP call. I doubt it's because they start the call using PushKit notification but rather a direct connection using VoIP certificate. Is that right?
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