Requires Standard or Premiere Voice editions
SIP Link lets you use Google Voice features, such as voicemail transcription, ring groups, and call forwarding, while maintaining your existing carrier and telephony routing solutions. This is done by connecting your existing carrier to Google through a set of certified Session Border Controllers (SBC).
SIP Link connects to your SBC over a TLS connection. Media sent over the SIP trunk is encrypted using DTLS or SDES. This connection takes place through the internet or direct peering.
Supported SBC vendors
We're currently testing more SBC vendors. Check back for updates. To be considered as an SBC vendor, please email SIP Link interop ([email protected]) with your SBC details.
SIP Link is currently available for the following SBC vendors. Use the integration notes to configure your devices. If you need additional help with the integration notes, contact your SBC vendor or carrier.
The minimum supported firmware versions are certified to work with SIP Link. Newer firmware versions that are a part of the same major firmware patch series are also certified.
Audiocodes
Hardware models |
|
---|---|
Minimum supported version | 7.20A.258.628 (Recommended version: 7.40A.250.262) |
Media encryption |
|
External interop documentation | Audiocodes interop |
Audiocodes hybrid gateway
Hardware models |
|
---|---|
Minimum supported version | 7.20A.258.628 (Recommend version: 7.4) |
Media encryption |
|
External interop documentation | Audiocodes hybrid interop |
Cisco
Hardware models |
|
---|---|
Minimum supported version | IOS XE 17.6.2 or IOS XE 17.6.4 |
Media encryption |
SDES |
External interop documentation | Cisco interop |
Oracle
Hardware models |
|
---|---|
Minimum supported version | 9.0 release or later |
Media encryption | SDES |
External interop documentation | Oracle interop |
Ribbon
Hardware models |
|
---|---|
Minimum supported version | 9.2 for SWe Core and 5400 models, 9.0.7 for SWe Edge and 1000/2000 models. |
Media encryption |
|
External interop documentation |
While we support all carriers, here are some of the carriers that have tested SIP Link with their network.
Carrier | Country |
---|---|
Bandwidth | Global |
Colt | Europe |
IPLAN | Argentina |
Orange | France |
Tata Communications | Global |
Telefonica | Global |
Telmex | LATAM |
Firewall Ports for SIP Link
Direction | Source IP Address |
Source Port | Destination IP address |
Destination Port |
Protocol |
---|---|---|---|---|---|
SBC to Voice | SBC Public IP | Your SBC Port | siplink.telephony.goog (signaling) | 5672 | TLS |
SBC to Voice | SBC Public IP | Your SBC Port | 74.125.39.0/24 (media) | 1024-65535 | DTLS/SDES (udp) |
Voice to SBC | siplink.telephony.goog | 5672 | Customer SBC Public IP (signaling); Your SBC Port | TLS | |
Voice to SBC | 74.125.39.0/24 | 1024-65535 | Customer SBC Public IP (media); Your SBC Port | DTLS/SDES (udp) |
Note: The TCP or UDP may change. For more info, read Voice connectivity requirements.
Install a TLS certificate certified for SIP Link
SIP Link accepts TLS certificates from the following Certificate Authorities (CAs):
- DigiCert
- Entrust DataCard
- GlobalSign
- GoDaddy
- Sectigo
TLS certificate requirements
TLS certificates must contain the fully qualified domain name (FQDN) of the SBC as the common name (CN), be 2,048 bits in size, and use RSA or ECDSA encryption. The domain of the CN in the certificate must match the Workspace domain.
Note: Wildcard certificates are not supported.
The TLS certificate and the trust chain from any one of the public CAs must be added to the TLS profile of the SBC along with the Google Root certificate.
To get the Google Root certificate:
- Download Google’s trusted root CAs.
- Extract GTS Root R1 (GTSR1).
- If required, extract the GlobalSign Root CA certificate.
- Upload the root certificates to your SBC keystore.
- Enable mutual authentication for the SBC.
- Once the certificate is installed, check for SIP Link ping options.
TLS connection requirements
Google Voice supports TLS version 1.2 or later. Voice requires clients to send the ClientHello with at least one supported cipher to succeed cipher negotiation. Your TLS connection must support the following cipher suite:
- TLS_ECDHE_ECDSA_WITH_AES_128_GCM_SHA256
- TLS_ECDHE_ECDSA_WITH_CHACHA20_POLY1305_SHA256
- TLS_ECDHE_RSA_WITH_AES_128_GCM_SHA256
- TLS_ECDHE_RSA_WITH_CHACHA20_POLY1305_SHA256
- TLS_ECDHE_ECDSA_WITH_AES_256_GCM_SHA384
- TLS_ECDHE_RSA_WITH_AES_256_GCM_SHA384
Set up SIP Link trunk between SBC & Voice
-
Sign in to your Google Admin console.
Sign in using your administrator account (does not end in @gmail.com).
-
In the Admin console, go to Menu
Apps
Google Workspace
Google Voice.
- Click SIP Trunks
Add a SIP trunk.
- Create a name for the SIP trunk
click Continue.
-
Click Add an SBC
and enter the following information:
- For SBC name, enter the name of the SBC.
- For Host name, enter the SBC host name that matches the CN on your TLS certificate. The domain must match the Workspace domain.
- For Model, enter the model of your SBC. Some models might not function properly with Voice. For details, go to SBC supported vendors & models.
- For TLS certificate, upload the TLS certificate signed by the trusted CA that’s associated with the SBC (must be under 5 MB).
- Click Save.
- (Optional) To add more SBCs, repeat the steps above. For high availability (HA) pairs, you must specify the active host name.
- Select the media encryption type for your SBCs. If you have multiple SBCs, they all must use the same encryption type. Voice accepts SDES and DTLS encryption.
- Click Continue.
- For Add a SIP header, copy the unique secret key and assign it
- Assign the SIP header name, X-Google-Pbx-Trunk-Secret-Key.
- Click Finish.
If you lose your secret key, you can find it on the SIP Trunk page by opening the SIP trunk and clicking Show secret key. When set up is complete make sure to read, Troubleshoot SIP Link setup.
Bandwidth Estimation for SIP Link
SIP Link over the Internet will need to account for the appropriate bandwidth capacity for the Internet connection.
SIP Link to User Call (Inbound Call) or User to SIP Link Call (Outbound Call)
Each SIP Link call to a user will take 84kbps(SIP Link side using g.711a/mu) + 50kbps (client side using Opus) = 134Kbps.
If we have 100 simultaneous calls then the bandwidth required is (134Kbps x 100) = 13400 Kbps or 13.4Mbps.
User to User Call (Intra Google Voice call)
Each User to User call will take 50kbps (client-A side using Opus) + 50kbps (client side using Opus) = 100Kbps.
Similarly, if we have 100 of such calls taking place, then 100 x 100Kbps = 10000Kbps or 10Mbps.
Note: The Internet connection will need to be monitored in real time to ensure sufficient capacity, or designed keeping in mind the number of calls of different types.
Related topics
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