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Bug 2446 - SIP calls/invites with camera active fails to respond to proxy auth request
: SIP calls/invites with camera active fails to respond to proxy auth request
Status: RESOLVED FIXED
Product: Chat & Call & SMS
SIP
: 4.0
: N800 other
: Medium normal (vote)
: ---
Assigned To: rtcomm@maemo.org
: sip-bugs
:
:
:
:
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Reported: 2007-12-01 03:51 UTC by James Pooton
Modified: 2008-02-12 13:33 UTC (History)
1 user (show)

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Description James Pooton (reporter) 2007-12-01 03:51:23 UTC 
BACKGROUND:  I'm using OS2008 beta on an N800 with an Asterisk server,
attempting to get g729/h263 video calls to work reliably with a PC client
(Eyebeam).  I'm almost there! However, I've hit a couple snags, one of which
*appears* to be rtcomm's fault.

STEPS TO REPRODUCE THE PROBLEM:

Initiate a SIP video/audio call *from* the N800 *with* the camera active
(extended).

EXPECTED OUTCOME:

SIP invite should be issued, followed by (if requested) an invite with proxy
authentication, resulting in successful call initiation.

ACTUAL OUTCOME:

Only the initial SIP invite is issued if the camera is active. If proxy
authentication is requested, a followup invite is never sent. This results in
an failed call attempt and even seems to unregister the N800 from the SIP
server quite often. 

If the camera is *not* active/extended (i.e. audio only call) and the call is
initiated from the N800 then a second invite *is* sent with proxy
authentication as requested and works successfully.

I'm attaching the SIP dialog to the end at the end here in both cases.

REPRODUCIBILITY:

Always


=========================================
SIP dialog WITH camera extended (failure)
=========================================

<--- SIP read from 205.118.58.12:64459 --->
INVITE sip:9000@voip.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.40:49477;rport;branch=z9hG4bKmj395ZHNmtjtc
Max-Forwards: 70
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=KZ5y7Npg737eS
To: <sip:9000@voip.ourdomain.com;user=phone>
Call-ID: 406edacd-1a15-122b-c2bd-00194fd4fcce
CSeq: 92105396 INVITE
Contact: <sip:testuser-mobile@205.118.58.12:64459;transport=udp>
User-Agent: Telepathy-SofiaSIP/0.4.1 sofia-sip/1.12.6work
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, UPDATE
Supported: timer, 100rel
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 511

v=0
o=- 3742772521436150005 4846668765947960776 IN IP4 192.168.2.40
s=-
t=0 0
m=audio 7078 RTP/AVP 96 18 8 0 13 97
c=IN IP4 192.168.2.40
b=RS:0
b=RR:0
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:97 telephone-event/8000
m=video 9078 RTP/AVP 34 96 97
c=IN IP4 192.168.2.40
b=RS:0
b=RR:0
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=2
a=rtpmap:96 H263-1998/90000
a=fmtp:96 QCIF=2
a=rtpmap:97 H263-N800/90000

<------------->
--- (15 headers 24 lines) ---
Sending to 205.118.58.12 : 64459 (NAT)
Using INVITE request as basis request - 406edacd-1a15-122b-c2bd-00194fd4fcce
www*CLI> 
<--- Reliably Transmitting (NAT) to 205.118.58.12:64459 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.40:49477;branch=z9hG4bKmj395ZHNmtjtc;received=205.118.58.12;rport=64459
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=KZ5y7Npg737eS
To: <sip:9000@voip.ourdomain.com;user=phone>;tag=as6df958e8
Call-ID: 406edacd-1a15-122b-c2bd-00194fd4fcce
CSeq: 92105396 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="voip.ourdomain.com",
nonce="5a1393af"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '406edacd-1a15-122b-c2bd-00194fd4fcce' in
32000 ms (Method: INVITE)
Found user 'testuser-mobile'
www*CLI> 
<--- SIP read from 205.118.58.12:64459 --->
ACK sip:9000@voip.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.40:49477;rport;branch=z9hG4bKmj395ZHNmtjtc
Max-Forwards: 70
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=KZ5y7Npg737eS
To: <sip:9000@voip.ourdomain.com;user=phone>;tag=as6df958e8
Call-ID: 406edacd-1a15-122b-c2bd-00194fd4fcce
CSeq: 92105396 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---

No further communication until the connection times out or the N800 become
unregistered.....


============================================
SIP dialog WITH OUT camera extended (success)
============================================

<--- SIP read from 205.118.58.12:64453 --->
INVITE sip:9000@voip.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.40:49479;rport;branch=z9hG4bKjva1568ZUeUrQ
Max-Forwards: 70
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105875 INVITE
Contact: <sip:testuser-mobile@205.118.58.12:64453;transport=udp>
User-Agent: Telepathy-SofiaSIP/0.4.1 sofia-sip/1.12.6work
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, UPDATE
Supported: timer, 100rel
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 323

v=0
o=- 7161779339802057598 7813639004219236037 IN IP4 192.168.2.40
s=-
t=0 0
m=audio 7078 RTP/AVP 96 18 8 0 13 97
c=IN IP4 192.168.2.40
b=RS:0
b=RR:0
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:97 telephone-event/8000

<------------->
--- (15 headers 15 lines) ---
Sending to 205.118.58.12 : 64453 (NAT)
Using INVITE request as basis request - 7b8f4631-1a17-122b-1aa0-00194fd4fcce
www*CLI> 
<--- Reliably Transmitting (NAT) to 205.118.58.12:64453 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
192.168.2.40:49479;branch=z9hG4bKjva1568ZUeUrQ;received=205.118.58.12;rport=64453
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>;tag=as64577a7c
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105875 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="voip.ourdomain.com",
nonce="7b7b56d7"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '7b8f4631-1a17-122b-1aa0-00194fd4fcce' in
32000 ms (Method: INVITE)
Found user 'testuser-mobile'
www*CLI> 
<--- SIP read from 205.118.58.12:64453 --->
ACK sip:9000@voip.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.40:49479;rport;branch=z9hG4bKjva1568ZUeUrQ
Max-Forwards: 70
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>;tag=as64577a7c
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105875 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
www*CLI> 
<--- SIP read from 205.118.58.12:64453 --->
INVITE sip:9000@voip.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.40:49479;rport;branch=z9hG4bKK53S71S3rQHBK
Max-Forwards: 70
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105876 INVITE
Contact: <sip:testuser-mobile@205.118.58.12:64453;transport=udp>
User-Agent: Telepathy-SofiaSIP/0.4.1 sofia-sip/1.12.6work
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, UPDATE
Supported: timer, 100rel
Proxy-Authorization: Digest username="testuser-mobile",
realm="voip.ourdomain.com", nonce="7b7b56d7", algorithm=MD5,
uri="sip:9000@voip.ourdomain.com;user=phone",
response="b0a7f6ae48df161cf90e9344fcd70875"
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 323

v=0
o=- 7161779339802057598 7813639004219236037 IN IP4 192.168.2.40
s=-
t=0 0
m=audio 7078 RTP/AVP 96 18 8 0 13 97
c=IN IP4 192.168.2.40
b=RS:0I> 
b=RR:0
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:13 CN/8000
a=rtpmap:97 telephone-event/8000

<------------->
--- (16 headers 15 lines) ---
Sending to 205.118.58.12 : 64453 (NAT)
Using INVITE request as basis request - 7b8f4631-1a17-122b-1aa0-00194fd4fcce
Found user 'testuser-mobile'
Found RTP audio format 96
Dynamic payload.
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 13
Found RTP audio format 97
Dynamic payload.
Peer audio RTP is at port 192.168.2.40:7078
Found description format iLBC for ID 96
Found description format G729 for ID 18
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format CN for ID 13
Found description format telephone-event for ID 97
Capabilities: us - 0x80100 (g729|h263), peer - audio=0x50c
(ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100
(g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3
(telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.40:7078
Looking for 9000 in pedoodles-dialplan (domain voip.ourdomain.com)
list_route: hop: <sip:testuser-mobile@205.118.58.12:64453;transport=udp>

<--- Transmitting (NAT) to 205.118.58.12:64453 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.2.40:49479;branch=z9hG4bKK53S71S3rQHBK;received=205.118.58.12;rport=64453
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105876 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9000@70.85.40.228>
Content-Length: 0


<------------>
    -- Executing [9000@pedoodles-dialplan:1]
Dial("SIP/testuser-mobile-0a0f3708", "SIP/testuser||Tt") in new stack
    -- Called testuser
    -- SIP/testuser-0a0fa868 is ringing
www*CLI> 
<--- Transmitting (NAT) to 205.118.58.12:64453 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.2.40:49479;branch=z9hG4bKK53S71S3rQHBK;received=205.118.58.12;rport=64453
From: <sip:testuser-mobile@voip.ourdomain.com>;tag=UcpZK4Np1ZNDS
To: <sip:9000@voip.ourdomain.com;user=phone>;tag=as16c7f9f6
Call-ID: 7b8f4631-1a17-122b-1aa0-00194fd4fcce
CSeq: 92105876 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:9000@70.85.40.228>
Content-Length: 0


<------------>
Comment 1 James Pooton (reporter) 2007-12-01 03:52:38 UTC 
Sorry, I would have made the traces attachements but didn't see that option
when creating the bug report initially. :/
Comment 2 Mikhail Zabaluev nokia 2007-12-03 14:05:30 UTC 
This looks vaguely familiar...
Try to set transport to UDP (or TCP, if the proxy has it) in advanced settings
of the SIP account.
Comment 3 James Pooton (reporter) 2007-12-03 18:25:49 UTC 
Sorry, I should have mentioned, I do have things set via UDP (rather then auto
or TCP.  I don't believe Asterisk has TCP support but I haven't checked lately.
 If it does I'll give that a shot.  

I'm starting to think it's somehow connected to IP addressing and NAT.  I'm not
sure what happens behind the scenes, but my posted tests were to directly
connected public server (no NAT) from an N800 behind a router (with NAT). Audio
only calls that originate from the N800 send both invites and connect.  If the
camera is out and the same call is placed only the first invite is received.

However, if the N800 is directly connected (no NAT) and the camera is out then
both invites are received.

I believe I have everything propertly configured for NAT usage and can post my
configuration if that might help.  I guess I would expect the audio only calls
to behave the same if I didn't.

Let me know if there is anything else you'd like to see/try.
Comment 4 Mikhail Zabaluev nokia 2007-12-04 12:20:45 UTC 
The best course of action is actually to wait for the OS2008 release, due out
soon, and retry with that. The last pre-OS2008 beta update is quite old, we've
made a lot of bugfixes since.
Comment 5 James Pooton (reporter) 2007-12-04 23:19:45 UTC 
Will do... Thanks!
Comment 6 Mikhail Zabaluev nokia 2008-02-12 13:33:56 UTC 
This should have been fixed by IT OS 2008 release 2.2007.50-2 or later.
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