In July 2018, Google Voice discontinued their XMPP-based integration and moved to a SIP-based method. These changes were discovered and documented mostly by the VoIP hobbyist community.
- Asterisk configuration example to use with Asterisk master branch (version 17 development). Contributed by naf.
- Google Voice XMPP support will go away in June (DSL Reports forum) - Early discussion and speculation on Google Voice's SIP implementation
- DSLReports thread on development of Asterisk updates for GV-SIP
- ObiTalk forum - Polycom/Obihai users sharing their experience and some useful information about how Obi devices connect to Google Voice using SIP
Google Voice uses these standards extending what is commonly implemented in a SIP UAC.
- RFC 5626 (SIP Outbound) - the single registrar/UA model, CR/LF keepalives
- RFC 3608 (Service Routes)
- RFC 3325, the section on P-Preferred-Identity header, which you get from the P-Associated-URI headers in the REGISTER reply. (The URI here is assigned by Google and is a Base 32-encoded concatenation of the username you register and a 20-digit ID associated with your account. You don't need to encode or decode this; just use it in the P-Preferred-Identity header.)
- draft-ietf-sipcore-sip-authn-02, Third-Party Authentication for Session Initiation Protocol, section 3: Authentication using the Resource Owner Password Credentials flow
On the media side, GV requires the client to present a full ICE implementation.
Google Voice implements rtcp-mux (RFC 5761).
- G.711µ
- Opus
| Proxy | Registrar | Used by | Notes |
|---|---|---|---|
| voice.telephony.goog:5061 (TLS) | voice.sip.google.com | GV Android app | Dial in E.164/E.123 format (e.g. +1NXXNXXXXXX) |
| alt#.voice.telephony.goog:5061 | # = 1..? | ||
| obihai.telephony.goog:5061 (TLS) | obihai.sip.google.com | Obihai devices | Registration generates an "ObiTalk Device" entry on the GV settings page |
| alt#.obihai.telephony.goog:5061 | # = 1..? |
REGISTER sip:obihai.sip.google.com SIP/2.0
Contact: <sip:[email protected];transport=tls>;obn=identifier;reg-id=1;+sip.instance="unique-ID"
Expires: 3600
To: <sip:[email protected]>
Call-ID: ...
Via: SIP/2.0/TLS 10.10.10.10:38250;alias;rport;branch=...
From: <sip:[email protected]>;tag=...
CSeq: 1 REGISTER
Max-Forwards: 70
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO
Supported: outbound, path
Authorization: Bearer username="me",realm="obihai.sip.google.com",token="an access token"
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TLS 10.10.10.10:38250;rport=38250;branch=...;received=...;alias
Service-Route: <sip:ENCODED-ROUTE:5060;uri-econt=ENCODED-ROUTE-PART-2;lr>
Service-Route: <sip:SECOND-ENCODED-ROUTE:5060;transport=udp;lr;uri-econt=PART-2>
Require: outbound
Contact: <sip:[email protected]:38250;transport=tls>;obn=identifier;reg-id=1;+sip.instance="unique-ID";expires=...
To: <sip:[email protected]>;tag=...
From: <sip:[email protected]>;tag=...
Call-ID: ...
CSeq: 1 REGISTER
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REGISTER, SUBSCRIBE, UPDATE
P-Associated-URI: <sip:[email protected]>
P-Associated-URI: <sip:[email protected]>
Content-Length: 0
Store the Service-Routes and the encoded P-Associated-URI. These are used later when establishing a dialog. Service-Routes become Route: headers (maintain the same order) and the encoded P-Associated-URI is used for the P-Preferred-Identity: header.
Outbound calls are placed over the established socket.
INVITE sip:[email protected] SIP/2.0
Route: <sip:ENCODED-ROUTE:5060;uri-econt=ENCODED-ROUTE-PART-2;lr>
Route: <sip:SECOND-ENCODED-ROUTE:5060;transport=udp;lr;uri-econt=PART-2>
P-Preferred-Identity: <sip:[email protected]>
Max-Forwards: 19
Via: SIP/2.0/TLS 10.10.10.10:38250;alias;rport;branch=...
From: <sip:[email protected]>;tag=...
To: <sip:[email protected]>
Call-ID: ...
CSeq: 2 INVITE
Contact: <sip:[email protected]:38250;transport=TLS>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO
Supported: outbound, path, replaces, 100rel
Content-Type: application/sdp
Content-Length: ...
(sdp)
SIP/2.0 100 Trying
...
SIP/2.0 183 Session Progress
...
(sdp)
The 183 response will include an SDP and Google Voice will start sending early media.
There may be a 180 Ringing response after this. If so, the endpoint should locally play ringback tone to the caller (differs from common practice defined in RFC 3960).
SIP/2.0 200 OK
...
(sdp)
The provisional responses and the 200 OK will contain Record-Route: headers which are used for in-dialog routing.
INVITE sip:[email protected]:38250;transport=tls SIP/2.0
Via: SIP/2.0/TLS 64.9.243.172:5061;branch=z9hG4bK...;rport
Via: SIP/2.0/UDP ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Via: SIP/2.0/UDP SECOND-ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Max-Forwards: 68
Record-Route: <sip:64.9.243.172:5061;lr;transport=tls>
Record-Route: <sip:ENCODED-ROUTE:5060;lr;transport=udp;uri-econt=PART-2>
Contact: <sip:+18005551212@ENCODED-DOMAIN:5060;transport=udp;uri-econt=PART-2>
To: <sip:[email protected]>
From: "caller" <sip:[email protected]:5060>;tag=...
Call-ID: ...
CSeq: 18524 INVITE
Allow: ACK, BYE, CANCEL, INVITE, UPDATE
Content-Type: application/sdp
Supported: 100rel
Privacy: none
P-Asserted-Identity: "caller" <sip:[email protected]>
P-Called-Party-ID: <sip:[email protected]>
Content-Length: 553
v=0
o=- 2125604025 1529079021317 IN IP4 74.125.39.26
s=SIP Call
c=IN IP4 74.125.39.26
t=0 0
a=ice-lite
a=ice-pwd:...
a=ice-ufrag:...
a=group:BUNDLE audio
a=fingerprint:sha-256 16:61:CE:09:B3:82:D2:81:DE:77:DB:B6:62:1C:CB:7E:D0:1B:F3:0B:D4:F7:D2:89:F1:74:35:45:2E:C3:FE:6E
a=setup:actpass
m=audio 19305 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=candidate:1 1 UDP 1 74.125.39.26 19305 typ host
a=candidate:2 1 UDP 2 2001:4860:4864:2::26 19305 typ host
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 64.9.243.172:5061;branch=z9hG4bK...;rport
Via: SIP/2.0/UDP ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Via: SIP/2.0/UDP SECOND-ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Record-Route: <sip:64.9.243.172:5061;lr;transport=tls>
Record-Route: <sip:ENCODED-ROUTE:5060;lr;transport=udp;uri-econt=PART-2>
To: <sip:[email protected]>
From: "caller" <sip:[email protected]:5060>;tag=...
Call-ID: ...
CSeq: 18524 INVITE
Server: ...
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 64.9.243.172:5061;branch=z9hG4bK...;rport
Via: SIP/2.0/UDP ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Via: SIP/2.0/UDP SECOND-ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Record-Route: <sip:64.9.243.172:5061;lr;transport=tls>
Record-Route: <sip:ENCODED-ROUTE:5060;lr;transport=udp;uri-econt=PART-2>
To: <sip:[email protected]>
From: "caller" <sip:[email protected]:5060>;tag=...
Call-ID: ...
CSeq: 18524 INVITE
Server: ...
Contact: <sip:[email protected]:38250;transport=TLS>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO
Content-Length: 0
SIP/2.0 200 OK
Via: SIP/2.0/TLS 64.9.243.172:5061;branch=z9hG4bK...;rport
Via: SIP/2.0/UDP ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Via: SIP/2.0/UDP SECOND-ENCODED-ROUTE:5060;branch=z9hG4bK...;econt=PART-2
Record-Route: <sip:64.9.243.172:5061;lr;transport=tls>
Record-Route: <sip:ENCODED-ROUTE:5060;lr;transport=udp;uri-econt=PART-2>
To: <sip:[email protected]>
From: "caller" <sip:[email protected]:5060>;tag=...
Call-ID: ...
CSeq: 18524 INVITE
Server: ...
Contact: <sip:[email protected]:38250;transport=TLS>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, PRACK, INFO
Content-Length: 424
v=0
o=- 11370442 1 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=audio 24624 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:24624
a=rtcp-mux
a=ptime:20
a=ice-ufrag:...
a=ice-pwd:...
a=candidate: (private IP)
a=candidate:... 1 UDP 2130706431 PUB.LIC.IP.ADDR 24624 typ host
ACK sip:[email protected]:38250;transport=tls SIP/2.0
...
- Google Voice apps for Android and iOS (Currently you must be in the "WiFi Beta" program to use the SIP calling feature)
- Obihai 200/202/212 (ATA), 1022/1032/1062/2062/2162/2182 (IP Phone)
- Asterisk "naf" fork - Now merged into the Asterisk source (Asterisk/master), this was a set of patches used to support Google Voice SIP. Use this sample config to connect Asterisk to Google Voice.
Submit an issue for contributions or corrections.