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Google Voice SIP Information

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.

Community Resources

Technical Details

SIP RFCs

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

Media requirements

ICE

On the media side, GV requires the client to present a full ICE implementation.

RTCP

Google Voice implements rtcp-mux (RFC 5761).

Codecs

  • G.711µ
  • Opus

Endpoints

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..?

Example Flow

Registration & establishing the TLS socket

Request (via proxy listed in table above; e.g. obihai.telephony.goog:5061/TLS)

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

Success Response

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

Request

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)

Responses

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.

Incoming call (INVITE)

Request

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

Responses

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

Request

ACK sip:[email protected]:38250;transport=tls SIP/2.0
...

Known Implementations

  • 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.

Contributions/Corrections

Submit an issue for contributions or corrections.

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