Failure of SIP Proxy Authentication
I'm developing a SIP user agent application that connects to an Asterisk server and tries to do an outgoing call. I'm using the NIST implementation of the JAIN SIP API.
When the application registers itself, a 401 (Unauthorized) response challenges it with a WWW-Authenticate header. The application inserts the Authorization header into the next REGISTER request. This time Asterisk returns a 200 (OK) response - the registration is successful.
When the application transmits an INVITE request, Asterisk responds with a 407 (Proxy Authentication Required) response. This time the response contains a Proxy-Authenticate header. My application sends an INVITE again, but this time with the Authorization header, upon which Asterisk responds with the same 407 (Proxy Authentication Required) response.
Here are the SIP messages that are transmitted ('>>' indicates outgoing messages; '<<' indicates incoming messages):
>>
REGISTER sip:10.0.84.30:5060 SIP/2.0
Call-ID: [email protected]
CSeq: 1 REGISTER
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKc7dd178d3d444ccc059a191e700fc8b73230
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Expires: 300
Content-Length: 0
<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKc7dd178d3d444ccc059a191e700fc8b73230;received=10.0.85.3
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
<<
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKc7dd178d3d444ccc059a191e700fc8b73230;received=10.0.85.3
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>;tag=as3c458716
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:[email protected]>
WWW-Authenticate: Digest realm="asterisk",nonce="6fbe5a68"
Content-Length: 0
>>
REGISTER sip:10.0.84.30:5060 SIP/2.0
CSeq: 2 REGISTER
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKffb0be254f93f61fa0dc7ac32b9078a43230
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Expires: 300
Authorization: Digest username="301",realm="asterisk",nonce="6fbe5a68",response="bc7075e8e241a4109dfa24d6ae95e78c",algorithm=MD5,uri="sip:10.0.84.30:5060",nc=00000001
Call-ID: [email protected]
Content-Length: 0
<<
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKffb0be254f93f61fa0dc7ac32b9078a43230;received=10.0.85.3
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
<<
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKffb0be254f93f61fa0dc7ac32b9078a43230;received=10.0.85.3
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:301@asterisk>;tag=as3c458716
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Expires: 300
Contact: <sip:10.0.85.3:5060>;expires=300
Date: Tue, 03 May 2011 06:42:33 GMT
Content-Length: 0
>>
INVITE sip:302@asterisk SIP/2.0
Call-ID: [email protected]
CSeq: 3 INVITE
From: <sip:301@asterisk>;tag=KOZWxg
To: <sip:302@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKaa0520efde83907b71d1f76315188c413230
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30:5060;lr>
Content-Type: application/sdp
Content-Length: 106
>>
v=0
o=- 3513393083 3513393083 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 40000 RTP/AVP 3
<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bKaa0520efde83907b71d1f76315188c413230;received=10.0.85.3
From: <sip:301@asterisk>;tag=KOZWxg
To: <sip:302@asterisk>;tag=as5de9ed83
Call-ID: [email protected]
CSeq: 3 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk",nonce="74986b64"
Content-Length: 0
>>
INVITE sip:302@asterisk SIP/2.0
CSeq: 4 INVITE
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:302@asterisk>
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK86f9dbdff9eeca422fbb67321dd45f7a3230
Max-Forwards: 70
Contact: <sip:10.0.85.3:5060>
Route: <sip:10.0.84.30:5060;lr>
Content-Type: application/sdp
Authorization: Digest username="301",realm="asterisk",nonce="74986b64",response="a08b8d7ce96cae00e7d334e132bf7358",algorithm=MD5,uri="sip:302@asterisk",nc=00000001
Call-ID: [email protected]
Content-Length: 106
>>
v=0
o=- 3513393083 3513393083 IN IP4 10.0.85.3
s=-
c=IN IP4 10.0.85.3
t=0 0
m=audio 40000 RTP/AVP 3
<<
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.85.3:5060;branch=z9hG4bK86f9dbdff9eeca422fbb67321dd45f7a3230;received=10.0.85.3
From: <sip:301@asterisk>;tag=2B3n8g
To: <sip:302@asterisk>;tag=as3c458716
Call-ID: [email protected]
CSeq: 4 INVITE
User-Agent: Asterisk PBX (switchvox)
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY
Contact: <sip:10.0.85.3:5060>
Proxy-Authenticate: Digest realm="asterisk",nonce="1bd30f50"
Content-Length: 0
The Authorization header is constructed in exactly the same way in both cases (same code that is executed). I'm using the request's request-URI for "digestURI". I've tried using a Proxy-Authorization header instead of an Authorization header, but the result is the same.
Can anyone see what I'm doing wrong? Thanks in advance.
Answer
For authenticating to a proxy (in other words you got a 407 Proxy Authentication Required you need a Proxy-Authorization header.
As RFC 2617 says, you construct this in the same way as you would an Authorization header.
You mention using the From URI in your question. RFC 2617 section 3.2.2 says you use the Request-URI (sip:302@asterisk). Watch out for the SIP-specific changes in RFC 3261 section 22.4.