In this post I like to highlight the availability to SIP gateway for Microsoft Teams. Microsoft announced the SIP gateway for compatible phones as general available in a recent Teams Blog post. This new service added and extending the Teams calling service to attach compatible SIP phones to your telephony deployment based on Microsoft Teams. In the following paragraphs I sum up the key aspects of the SIP gateway. Hereinafter, I do not explain Teams PSTN calling options.
Available features for compatible SIP phones [16.12.2021]
By the time of writing this article the following features are mentioned in the documentation to be supported:
- Make and receive calls
- Hold and resume
- Mute and unmute
- Multiple simultaneous calls by using hold and three-party confernce call
- DTMF (Dual-tone multi-frequency)
- Teams meetings dialin
- Cold call transfer
- Consultative call transfer
- Local call forwarding
- Listen to voicemail
- Message waiting indicator (MWI)
- Do not disturb only for the SIP phone
- User sign-in/-out
- Skype for Business phones which connect via a third-party gateway can use the SIP gateway beyond 2023
In case a sip phone should be used the Teams users must have a phone number with PTSN calling and a suited licenses assigned (e.g. E1, phone system license for Direct Routing or else).
The Teams calling policy must be enabled for SIP phones and assigned to the users for SIP phones.
Moreover, your IT infrastructure must allow SIP phone traffic as well (firewall, proxy, …).
I’m not going to explain the configuration here, because it is very well documented on Configure SIP Gateway – Microsoft Teams | Microsoft Docs.
Administration and monitoring
SIP phones can be monitored and administered to a certain extend via Teams Admin Center.
Conclusion, opinion and summary
SIP gateway for Microsoft Teams Voice deployments are a good opportunity to simplify the deployment because you do not need to deploy your own one or connect all SIP phones to a SBC. Furthermore you could keep existing legacy SIP phones, if compatible.
As of now there is no support for analog device gateways to register analog endpoints via the analog device gateway like a SIP endpoint, e.g. in case of Teams Calling Plans and the need for a few analogs like a doorbell or fax machine. But one step after another.
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You are welcome and thank you for your feedback.
– Dialing out: 10 seconds is a long time for dial out to start but it can be influenced by many variables, e.g. SIP device connectivity (especially latency) to the SIP GW, end-to-end network path from SIP device to Teams Phone System backend to the SBC to the PSTN SIP carrier, if there are any firewall/proxies… inbetween, … Are there any local device settings regarding dialout timers, e.g. to wait x seconds until the device accepts the entered number to dial out (not relevant if you force the device to dial out by pressing the dial button instead of “waiting”)
– Music on hold: Generally speaking, this can often be resolved by SBC configuration adjustments, e.g. with a prerecorded tone file and suited SBC hold configuration (on an Audiocodes SBC). For Audiocodes SBCs, there was an update in their Audiocodes Mediant VE Teams Direct Routing deployment guide which covers how to configure Music on hold for consultative call transfers in Microsoft Teams (not the same).
Have a good time
Just wanted to say thank you for your quick reply! We finally managed to fix the dialing out issue by fine-tuning our firewall policy and adding QoS on the signaling traffic, we are now at 2-3 seconds which is “normal”.
Regarding the Music on Hold problem, I’m not sure we will have a solution for this as we are using provider hosted SBCs and they don’t seem to want to implement music on hold on them.
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thanks a lot for this new blog article and for all the things you share!
I did configure Cisco IP Phones and connected them to the SIP Gateway successfully, nevertheless, I found the following “issues”:
– Dialing out takes a lot of time (at least 10 seconds until I get the ringback tone) and I am connected on the SIP gateway of my region.
– Music on Hold is not working when putting a call from the PSTN on hold on the phone.
Maybe the 2 above mentioned issues come from a configuration mistake on my direct routing setup, but I would like to know if I’m the only one experiencing this with SIP Gateway.
By the way, if the user changes his password, the SIP phone will get logged out (same behavior as with Teams certified phones). I was hoping it would not be the case… but I can accept this one.
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