Polycom Teams Calls Transfers Not Completing when connected to Meraki Switches

[AnthonyCSheehy]

We seem to have a peculiar problem of Teams devices apparently not receiving their SIP Invites for Call Transfers whilst phones are connected to Meraki switches. 

We can take them to another site where, connected to Cisco 2960x switches, the call transfer completes without a problem. 

What are we doing wrong and how can we fix it?

[rhbirkelund]

Disclaimer: I don't know SIP or Polycom.

Best guess would be multicast. Try looking at IGMP, and perhaps create a SVI on the vlan with SIp devices, on the switch closest as a core switch.

[cmr]

I'd agree with @rhbirkelund, on the vlan that the devices are on, create an SVI on a switch, or edit the existing and enable the IGMP snooping querier.  It is not enabled by default.

[Leosambrana]

IGMP snooping is enabled by default on MS. 

This prevents the switch from sending multicast traffic to hosts who are not yet joined with the proper multicast group. IGMP Snooping can be disabled under the Switch > Switch Settings page in the Dashboard.

This might be your case.

[AnthonyCSheehy]

Thanks. I am attempting this and will report back. 

PS. I'm not full time voice support hence why I get back slowly.

[PhilipDAth]

Do the two sites use different firewalls?

Does the firewall for the site show any traffic being blocked?

I don't think this is likely to be a switch issue.

[AnthonyCSheehy]

No. It's on a WAN link, but there's no firewall in-between. The transfer fails to happen in-site as well as when attempting to transfer to the next site. There's no such failure when transferred on the mail site.

The only difference between the two sites is that they are on Meraki switches, and the main site is on Cisco switches.

[PhilipDAth]

Anything in the Meraki event log?

[PhilipDAth]

I think you are going to need to do some packet captures to solve this.

Do a packet capture where the WAN plugs into the switch.  Do you see the SIP invite then?

Do a packet capture where the phone plugs into the switch.  Do you see the SIP invite then?

[IvanJukic]

For troubleshooting Voice, VoIP, Audio etc we treat it like an application. E.g. user can’t receive emails. If the end device can make calls, but not receive only during transfer. It would be blocked somewhere. So, need to establish if SIP is making its way through to the end device when trying to transfer.

Suggest the following between phones at the site. i.e. ‘internal calls’.

First, check VLANs, ACLs, Port Forward/NAT etc towards the port connected to the phones.

Second, need to run a pcap on the port that is connected to the phone. Make some calls from the phone. Then try the transfer to the phone.

Cheers,
Ivan Jukić