vaibhav3002 / sipdroid Goto Github PK

Vonage is SIP at it's core but it uses their own home-grown
authentication system.  Would it be possible to integrate Vonage
support into Sipdroid?

Vonage released the source for Vonage Talk awhile back, and it is
available via the GPL

Vonage Talk can be found at:
http://alpha.vonage.com/vtalk

You can download the source at Sourceforge.net:
http://sourceforge.net/projects/vtclient

As a Vonage user, I think this would be a fantastic feature. 

When connected via Sipdroid to an asterisk server, the device is able to
authenticate (as long as the realm is set to the FQDN and NAT is enabled).

I am not however able to establish a call path. SIP Debug below:

<--- SIP read from 203.206.171.85:2371 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK66842
Max-Forwards: 70
To: <sip:[email protected]>;tag=as2a431003
From: <sip:[email protected]>;tag=z9hG4bK10903280
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
<--- SIP read from 203.206.171.85:2371 --->
REGISTER sip:pbx.purpleoranges.com.au SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK22106
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK57143023
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 203.206.171.85 : 2371 (NAT)
voip001*CLI> 
<--- Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK22106;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK57143023
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
voip001*CLI> 
<--- Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK22106;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK57143023
To: <sip:[email protected]>;tag=as23ddd50b
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="pbx.purpleoranges.com.au",
nonce="6848d84b"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms
(Method: REGISTER)
Reliably Transmitting (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK6b79a498;rport
From: "asterisk" <sip:[email protected]>;tag=as0c668be1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
REGISTER sip:pbx.purpleoranges.com.au SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK22106
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK57143023
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Authorization: Digest username="4004", realm="pbx.purpleoranges.com.au",
nonce="6848d84b", uri="sip:pbx.purpleoranges.com.au", algorithm=MD5,
response="fd04918adbb763505429517be16eb2fd"
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Using latest REGISTER request as basis request
Sending to 203.206.171.85 : 2371 (NAT)
voip001*CLI> 
<--- Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK22106;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK57143023
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
voip001*CLI> 
<--- Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK22106;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK57143023
To: <sip:[email protected]>;tag=as23ddd50b
Call-ID: [email protected]
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Expires: 3600
Contact: <sip:[email protected]>;expires=3600
Date: Sat, 16 May 2009 20:34:46 GMT
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms
(Method: REGISTER)
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 152
Content-Type: application/sdp

v=0
[email protected] 0 0 IN IP4 127.0.0.1
s=Session SIP/SDP
c=IN IP4 127.0.0.1
t=0 0
m=audio 21000 RTP/AVP 8
a=rtpmap:8 PCMA/8000

<------------->
--- (12 headers 7 lines) ---
Sending to 203.206.171.85 : 2371 (NAT)
Using INVITE request as basis request - [email protected]
voip001*CLI> 
<--- Reliably Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK74158;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK72602749
To: <sip:[email protected]>;tag=as48c591d5
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="pbx.purpleoranges.com.au",
nonce="19b3cce5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms
(Method: INVITE)
Found user '4004'
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Expires: 3600
User-Agent: mjsip stack 1.6
Proxy-Authorization: Digest username="4004",
realm="pbx.purpleoranges.com.au", nonce="19b3cce5",
uri="sip:[email protected]", algorithm=MD5,
response="493638183226677a8c6d845591a1c4b8"
Content-Length: 152
Content-Type: application/sdp

v=0
[email protected] 0 0 IN IP4 127.0.0.1
s=Session SIP/SDP
c=IN IP4 127.0.0.1
t=0 0
m=audio 21000 RTP/AVP 8
a=rtpmap:8 PCMA/8000

<------------->
--- (13 headers 7 lines) ---
voip001*CLI> 
<--- Transmitting (NAT) to 203.206.171.85:2371 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK74158;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK72602749
To: <sip:[email protected]>;tag=as48c591d5
Call-ID: [email protected]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>;tag=as48c591d5
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #1 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK6b79a498;rport
From: "asterisk" <sip:[email protected]>;tag=as0c668be1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (NAT) to 203.206.171.85:2371:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK74158;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK72602749
To: <sip:[email protected]>;tag=as48c591d5
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="pbx.purpleoranges.com.au",
nonce="19b3cce5"
Content-Length: 0


---
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>;tag=as48c591d5
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #2 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK6b79a498;rport
From: "asterisk" <sip:[email protected]>;tag=as0c668be1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Reliably Transmitting (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK27d906ec;rport
From: "asterisk" <sip:[email protected]>;tag=as65403590
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #2 (NAT) to 203.206.171.85:2371:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK74158;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK72602749
To: <sip:[email protected]>;tag=as48c591d5
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="pbx.purpleoranges.com.au",
nonce="19b3cce5"
Content-Length: 0


---
voip001*CLI> 
<--- SIP read from 203.206.171.85:2371 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>;tag=as48c591d5
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #3 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK6b79a498;rport
From: "asterisk" <sip:[email protected]>;tag=as0c668be1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #1 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK27d906ec;rport
From: "asterisk" <sip:[email protected]>;tag=as65403590
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #4 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK6b79a498;rport
From: "asterisk" <sip:[email protected]>;tag=as0c668be1
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog
'[email protected]' Method: OPTIONS
Retransmitting #2 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK27d906ec;rport
From: "asterisk" <sip:[email protected]>;tag=as65403590
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Retransmitting #3 (NAT) to 203.206.171.85:2371:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK74158;received=203.206.171.85;rport=2371
From: <sip:[email protected]>;tag=z9hG4bK72602749
To: <sip:[email protected]>;tag=as48c591d5
Call-ID: [email protected]
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="pbx.purpleoranges.com.au",
nonce="19b3cce5"
Content-Length: 0


---
voip001*CLI> sip set 
<--- SIP read from 203.206.171.85:2371 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK74158
Max-Forwards: 70
To: <sip:[email protected]>;tag=as48c591d5
From: <sip:[email protected]>;tag=z9hG4bK72602749
Call-ID: [email protected]
CSeq: 1 ACK
User-Agent: mjsip stack 1.6
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Retransmitting #3 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK27d906ec;rport
From: "asterisk" <sip:[email protected]>;tag=as65403590
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog
'[email protected]' Method: OPTIONS
Retransmitting #4 (NAT) to 203.206.171.85:2371:
OPTIONS sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 202.131.92.18:5060;branch=z9hG4bK27d906ec;rport
From: "asterisk" <sip:[email protected]>;tag=as65403590
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 16 May 2009 20:34:48 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


---
Really destroying SIP dialog
'[email protected]' Method: OPTIONS

What steps will reproduce the problem?
1. Visit http://code.google.com/p/sipdroid/wiki/FAQ

What is the expected output? What do you see instead?
I would like to see which codecs are supported but instead nothing like
that is shown. A working sip.conf entry for Asterisk would be great. If it
works with SIPes then it would be great to know what they use.

What version of the product are you using? On what operating system?
[Not applicable]

Please provide any additional information below.
I have gotten it to work with a-law in Asterisk. But I can only hear calls
in one direction.

[other sip-phone] -> [sipdroid]

What steps will reproduce the problem?
1. Calling outbound on EDGE network (AT&T)

What is the expected output? What do you see instead?
I can call fine over WiFi using pbxes and 12voip but when I switch to EDGE and 
dial, the call does 
not go through. It appears to be trying to call, but then ends the call after 
15-20 seconds of not 
ringing. Also, my call log on pbexs shows no calls being made (but, of course, 
shows the calls I 
made over WiFi)

What version of the product are you using? On what operating system?
Siproid 0.9, Android 1.5

Please provide any additional information below.

Either register with Asterisk or pbxes.org
I can call out but can not receive incoming call.

What steps will reproduce the problem?
1. Utilize Haykuro HTC Build
2. Configure to connect to sip server

What is the expected output? 
register with sip server

What do you see instead?
stays in idle without attempt to connect to sip server

What version of the product are you using? On what operating system?
haykuro 5.0.2H build

Please provide any additional information below.
if i deactivate 3G/HSD and utilize on EDGE or WIFI, it works perfectly.

What steps will reproduce the problem?
1. start sipdroid
2. try to find a close button

What is the expected output? What do you see instead?
I'd love to be able to actually close sipdroid. It's nice that I'm able to
run it in the background, but a close option would come in handy

What version of the product are you using? On what operating system?
0.9 sipdroid, 1.5 android adp

What steps will reproduce the problem?
1. Enter sip settings for a local asterisk server
2. Attempt to register
3. Packet capture shows Sent-by Address: 127.0.0.1 and contact 127.0.0.1
(localhost)

What is the expected output? What do you see instead?

Can it not send the ip of the device? Perhaps it expects some received
address and received port processing?

What version of the product are you using? On what operating system?

1.5 official on adp 1

Please provide any additional information below.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Set up everything.
2. Call your G1 phone from a SIP client, it shows up on the screen but does
NOT ring.
3. Do not see option either in the sipdroid or the phone to correct this.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?
G1 running Cupcake.


Please provide any additional information below.

There is no source code availabe in the Subversion.

I can't connect to the Voip-Server.
I always get a Timeout.

After the Installation the first time it was connected.
After I changed the settings no connection any more.

I reinstalled sipdroid but I have still the same problem.

In a trace with Wireshark on the voip-server I couldn't see any traffic
between the phone an the server.

The installed Version is 0.9

Firmware-Version 1.5

Kernel-Version 2.6.27-00392-g8312baf
android-build@apa27 #72

Build-Number
CRB17

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Try BT
2.
3.

What is the expected output? What do you see instead?
The mic don't work, instead the mic on the g1 is active. Also the volume in
the BT set is too low.

What version of the product are you using? On what operating system?
Android 1.5 jesusfreke, Sipdroid 0.9

Please provide any additional information below.

What steps will reproduce the problem?
1. register on voipbuster
2. call anybody

What is the expected output? What do you see instead?
No sound at all. Call & answer are OK, but no sound at all with this
provider (voipbuster). I tried in 3G & Wifi, same result.
This account is OK with my PC.
I tried another temporary provider (Gizmo5) with SipDroid and everything is
perfect. 
Why is Gyzmo5 ok, and voipbuster ko ?

What version of the product are you using? On what operating system?
version 0.9 + cupcake Dude 0.8

What steps will reproduce the problem?
1. Make a call
2. Capture and examine RTP packets.
3.

What is the expected output? What do you see instead?

One RTP packet should contain 20ms of audio or so, at least this is what
asterisk does.

What version of the product are you using? On what operating system?

0.9 on Android 1.5

Please provide any additional information below.

The RTP payload is to be 1024 bytes in size, when asterisk sends 160 byte
payloads.

It would be really nice if you could set up rules on when to use SIP.

Something like All numbers that doesn't match +45* should use SIP.
(I'm from denmark)

Or when I call Peters cellphone, it should make a regular call.

We have tried to use Sipdroid with our SoftSwitch but it can not
authenticate the REGISTRATION because the branch id is not changed by
Sipdroid when resending REGISTRATION request with authentication
information and so our SoftSwitch identifies it as the same transaction as
the original REGISTRATION request and simply sends back the stored reply.

For the first REGISTRATION request from Sipdroid we answer a 401
Unauthorized and that reply terminates the transaction, so the next
REGISTRATION request with authentication information must have a different
branch id!

Original request:
REGISTER sip:deverto.com SIP/2.0
Via: SIP/2.0/UDP
127.0.0.1:5060;rport=5060;branch=z9hG4bK47254;received=212.40.113.58
Max-Forwards: 70
Contact: <sip:[email protected]>
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK12344022
Call-ID: [email protected]
CSeq: 1 REGISTER
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0

Our reply:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5060;rport=5060;branch=z9hG4bK47254;received=212.40.113.58
To: <sip:[email protected]>;tag=6f65be3c
From: <sip:[email protected]>;tag=z9hG4bK12344022
Call-ID: [email protected]
CSeq: 1 REGISTER
WWW-Authenticate: Digest
realm="deverto.com",algorithm=MD5,nonce="4a0420aab3423b1951c080608ef08768ea44ca2
5",qop="auth",o
paque="",stale=false
Content-Length: 0

----------------
New REGISTER request from Sipdroid:

REGISTER sip:deverto.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK47254
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK12344022
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Authorization: Digest username="huhha", realm="deverto.com",
nonce="4a0420aab3423b1951c080608ef08768ea44ca25", uri="sip:d
everto.com", algorithm=MD5, opaque="", qop=auth,
response="6fd1a78a6ed447d6c3a9baf077013d97"
Content-Length: 0

Notice that the branch id is the same.

If the branch id starts with the magic cookie "z9hG4bK" (as in this case),
then it must be unique for every transaction.

See rfc 3261 for further details.

Well according to the Sipdroid Users group, it's more a feature request
than an issue.

Here is the background. In france we have 2 ISP that provide a SIP account
in the base package. That's near 8 millions users who's got a SIP account
(95% of them don't know that). Those SIP/VOIP account have free
international call for more than 90 country, but only for "home call".

It would be very interresting in sipdroid, in "preferred call type", to
have another item called : "sip only on home number"

So that the cheapest way would be choose everytime.

Perhaps in another country, the "sip only on mobile number" would be more
interresting for the same reason.

First of all: Great job guys keep up with your Sipdroid development. This 
is great application which filled the VoIP gap in Android world!! 

What steps will reproduce the problem?
1. Call into any conference
2. Try to mute microphone
3.

What is the expected output? What do you see instead?

- That should be easy to correct. During conversation mute button should 
mute microphone instead of holding on conversation. I tried other 
applications if I would be able to mute microphone with Sound Manager. It 
looks like it is not exposed. This option within Sipdroid is very critical 
during conference, when you don't want to disturb with noices.


What version of the product are you using? On what operating system?
- Sipdroid 0.9 (Apr 27)
- Android 1.5 Cupcake

Please provide any additional information below.

In r86 of trunk/src/org/zoolu/sip/address/SipURL.java, the SipURL(String
hostname, int portnumber) constructor calls init(null, hostname,
portnumber). On line 93 within init(String username, String hostname, int
portnumber), there is no test to see if username is null, so the
username.indexOf call can throw a NullPointerException.

First off, I am in the US, using version 1.5.  I believe my G1 phone was 
recently updated with the latest software.  

I've tried both, download from the market as well as download from this 
site. Both times I am getting the same results:

1. The screen says: "called party address"; is there something I am 
suppose to be input here?

2. When I do input anything; phone number, sip extension address, etc, I 
get "no suitable data network ready"

I have tried using the 3g and edge networks.  As well as my wireless at 
home.

Any advice would be greatly appreciated.

Nube here, 

Does anyone know what the settings need to be on PBXes.org to make Gizmo work?

Thanks in advance

What version of the product are you using? On what operating system?
v0.9 on Android Cupcake.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1.Download Sipdroid-0.9-full.apk
2.Install on T-mobile G1 (HTC Dream)
3.

What is the expected output? What do you see instead?

Application install unsuccessful

What version of the product are you using? On what operating system?
Firmware 1.1
Kernel 2.6.25-01845 Android build@apa27 RC33 126986

Please provide any additional information below.
contactme at android(at)ourhouse.ch

What steps will reproduce the problem?
1. Download sipdroid version 0.9-full
2. Enable the setting: Settings/Applications/Unknown Sources
3. use any file manager to install the apk file

This series of steps launches the installation procedure correctly.
However, installation ends unsucceffully.

0.9-full T-mobile G1 Android RC33

Thank you\ Kudret

In my companies network I'm unable to connect to SIP.
I always have the orange dot in the status bar.

It would be really nice to see some more detailed connection status to see
what the actual problem is.

If possible, it would be REALLY helpful if multiple codec support was
included as there are a couple codecs that would be much better for a phone
like this. Or even set a fallback codec, however that is not even close to
mattering...

The codecs that would be EPIC to see supported are:

ILBC - Internet Low Bitrate Codec (Good for edge, or low signal)
GSM  
G729
G711

Thanks!

What steps will reproduce the problem?
1. dial other person
2. listen

What is the expected output? What do you see instead?
The other person can hear me and the audio quality seems to be very good
(woohoo!). I can't hear the other person talking at all

What version of the product are you using? On what operating system?
Sipdroid 0.9 + the official ADP 1.5 Image with current radio fw

Please provide any additional information below.
Tried using sipgate.de directly and pbxes.com (with sipgate)

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. make a call to sip line, for example [email protected]
2. go to the call log, attempt to call the same address again
3. the address is converted to a set of digits (probably decimal
representation of characters)

What is the expected output? What do you see instead?
I expect it to repeat the call to [email protected] instead an
attempt is made to call a bunch of digits which are not known as either a
sip address or a valid phone number.

What version of the product are you using? On what operating system?
Android Cupcake 1.5

Please provide any additional information below.
The normal behaviour that we should see is the same as with regular phone
numbers, i.e. anything in the call log if clicked on or pressed (or
caressed depending on your preference) should invoke the expected action of
making another call to the same address.  It is not clear if the sipdroid
picks up the digits instead of an address or if the Android software does this.

outgoing calls with pbxes and sipgate works great. but incoming call via
pbxes doesn't work. can someone help me?

I installed both the market version 0.9.4 and the lastest Google Code
version 0.9.5-full. In neither case was I able to make it register to any
of these: FreePBX on the local LAN; pbxes.org aka pbxes.com; voip.ms. This
is however not a serious problem as the softwaerw is still being developed
and I'm sure bugs will be found and fixed.

THE REAL PROBLEM I am reporting here is that this program is apparently
dependent, by design, on pbxes.org. The documentation suggests this in
several places.

This dependence on a specific website is not good for a two reasons.

First, if the software depends on pbxes.org, then the developers will have
a lesser incentive to make it use standard SIP. Nonstandard behavior will
creep in and will remain undetected and become permanently embedded into
the design of the software.

Second, pbxes.org itself has some problems. It uses some sort of
nonstandard Adobe Flash code that always yields a strange error "Error
loading configuration file variables.txt?aldope=73939". Please don't make
your software dependent on a non-functioning web site.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Configure the UAC to use a SIP registrar that sets the qop parameter in
WWW-Authenticate/Proxy-Authenticate headers in the 401/407 reply.
2. Initiate registration process


What is the expected output? What do you see instead?
A WWW-Authenticate/Proxy-Authenticate header with cnonce and nc parameters

What version of the product are you using? On what operating system?
svn trunk in emulator

Please provide any additional information below.

The missing cnonce and nc values makes the registration process fail with
some SIP registrars. These parameters are mandatory if the qop parameter is
set according to section 3.2.2 of RFC 2617.

The example below shows that qop="auth" is set in the 401 reply and that
the following REGISTER is missing the nc and cnonce parameters in the
Authorization header which results in a 400 Bad Request reply from the
registrar.

The same behavior is observed with Proxy-Authenticate/407 transactions.

---

REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK00429
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK92140648
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0


SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5060;rport=33234;branch=z9hG4bK00429;received=192.168.1.10
To: <sip:[email protected]>;tag=9e3d5c42014a5041c27e22f1f36bbce6.462c
From: <sip:[email protected]>;tag=z9hG4bK92140648
Call-ID: [email protected]
CSeq: 1 REGISTER
WWW-Authenticate: Digest realm="example.com",
nonce="4a12b95f3b08eb9291a51b7c341446194a705efb", qop="auth"
Content-Length: 0


REGISTER sip:example.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK00429
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK92140648
Call-ID: [email protected]
CSeq: 2 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Authorization: Digest username="foo", realm="example.com",
nonce="4a12b95f3b08eb9291a51b7c341446194a705efb", uri="sip:example.com",
qop=auth, response="9d11f6ae2adfca089b596d8769b09ed2"
Content-Length: 0


SIP/2.0 400 Bad Request 
Via: SIP/2.0/UDP
127.0.0.1:5060;rport=33234;branch=z9hG4bK00429;received=192.168.1.10
To: <sip:[email protected]>;tag=9e3d5c42014a5041c27e22f1f36bbce6.462c
From: <sip:[email protected]>;tag=z9hG4bK92140648
Call-ID: [email protected]
CSeq: 2 REGISTER
Content-Length: 0

---

Patch:

diff -r 5f03464e13bb src/org/zoolu/sip/authentication/DigestAuthentication.java
--- a/src/org/zoolu/sip/authentication/DigestAuthentication.java        Tue
May 19 14:49:15 2009 +0200
+++ b/src/org/zoolu/sip/authentication/DigestAuthentication.java        Tue
May 19 16:17:28 2009 +0200
@@ -5,6 +5,7 @@ import org.zoolu.sip.header.ProxyAuthori
 import org.zoolu.sip.header.ProxyAuthorizationHeader;
 import org.zoolu.sip.header.WwwAuthenticateHeader;
 import org.zoolu.tools.MD5;
+import org.zoolu.tools.Random;

 /**
  * The HTTP Digest Authentication as defined in RFC2617. It can be used to i)
@@ -53,6 +54,14 @@ public class DigestAuthentication {
                init(method, ah, body, passwd);
                this.uri = uri;
                this.qop = qop;
+               if (qop != null && cnonce == null) {
+                       this.cnonce = Random.nextHexString(16);
+                       /*
+                        * A new cnonce is generated for every new instance so 
+                        * no need to keep track of the nc value.
+                        */
+                       this.nc = "00000001";
+               }
                this.username = username;
        }

@@ -119,6 +128,8 @@ public class DigestAuthentication {
                        ah.addQopParam(qop);
                if (nc != null)
                        ah.addNcParam(nc);
+               if (cnonce != null)
+                       ah.addCnonceParam(cnonce);
                String response = getResponse();
                ah.addResponseParam(response);
                return ah;

What is the expected output? What do you see instead?
A call from Gizmo5 online voip softphone is expected to arrive at Sipdroid
on Android Emulator, but instead the call goes to a voice mail box

What version of the product are you using? On what operating system?
I'm using the latest version of Sipdroid, running on an Android 1.5
emulator, on Windows XP platform.

Please provide any additional information below.
I have two PC's, both have internet connection, PC1 has Sipdroid running on
Android 1.5 emulator, PC2 is logged in to Gizmo5 browser-based voip
softphone, located here http://gizmo5.com/pc/.

If PC1 calls PC2 (Sipdroid to Gizmo5 online softphone), the browser phone
rings, when answered, the call is a success, both parties can hear each
other, except for a couple of seconds of delay

However, if PC2 calls PC1 (Gizmo5 online softphone to Sipdroid), the call
never reaches Sipdroid. It is not even a notification problem, the call
never really reach Sipdroid, even if you are guarding it every second,
neither when an outgoing call has just ended seconds ago. 

What you hear from Gizmo5 softphone is just a voice-recorded reply telling
you that you are at the voice mailbox, while Sipdroid just sits there as if
nothing happened.

What should I do to solve this problem? Do I need to modify the code? How?

I also couldn't register at pbxes.org, it keeps telling me incorrect
username and password when I newly register, so I couldn't make an account
successfully, and I can't try it there to see if the same problem exists in
a different sip provider. Any help about this? Thanks!

I think it could be useful to be able to configure a custom prefix number
that is always used when calling.
The first use I see is for hiding the phone number, in France for my sip
provider I need do dial 3651 before the contact number.
A french mobile provider "Bouygues Telecom" also use 777 as a prefix for
some special free calls.
I'm sure there's a lot of possible uses for this option.

SIPDROID 0.9.5 won't register to my asterisk

What steps will reproduce the problem?
1. Try wifi/3g/edge and can not see any registering traffic coming to the 
asterisk
3.

What version of the product are you using? On what operating system?
I am runing JF1.5.1ADP build, and sipdroid 0.9.5
the sipdroid keeps saying 401 Registration failed
but when I checked the asterisk, the sipdroid never connect to it.
looks like sipdroid never sent out register request to the server

Please provide any additional information below.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Toggle the flag to prefer SIP when available.
2. Try to dial any normal number, not SIP.
3. It appears to dial but nothing happens.

What is the expected output? What do you see instead?
Have to unset the SIP flag to use the phone normally.

What version of the product are you using? On what operating system?
G1 CUpcake 1.5

Please provide any additional information below.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Arrange incoming call
2.
3.

What is the expected output? What do you see instead?
Audible ring signal / The call is visually displayed.

What version of the product are you using? On what operating system?
Sipdroid 0.9, Andriod 1.5 (jesusfreke)

Please provide any additional information below.

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?


Please provide any additional information below.
i think its because it 1.5 build or that 3g is shown as h

Outgoing SIP calls have no dial tone.

And the called port comes on line rather unexpected because there is no
audible clue that the call is connected now. A little click noise would
help a lot there.

It is probably possible for the provider to inject a dial tone for me.
But mine does not (sipgate.de) in this case the handset should do it.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. Save a SIP URI such as [email protected] as one the phone numbers for a
contact.
2. Attempt to initiate a SIP call to the contact by tapping the "Text
Mobile" option for the SIP URI and selecting the "Call over IP" option.
3.

What is the expected output? What do you see instead?
Instead of calling the SIP URI, the entire URI is converted into numerals
using the keypad mapping (ABC -> 2, DEF -> 3, etc). Thus, instead of making
a sip call to [email protected], sipdroid calls 873777699747266.

What version of the product are you using? On what operating system?
Sipdroid 0.9.4-full, ADP1 v1.5

Please provide any additional information below.

hi

On my htc Dream with cupcake, I've setting sipdroid with my asterisk
server, all is fine but when I receive a call it's display on the android
screen, I can answer, but the phone don't ring.

1. Use sipdroid with settings(wlan,edge,3g) turned off
2. Make phonecall (ie voicemail) and attempt to either slide or menu-enable
the dialpad
3. Dialpad shows if you select from menu, but DTMF numbers dont work, slide
doesn't either.

What version of the product are you using? On what operating system?
sipdroid 0.9.5 on cupcake 1.5 

Please provide any additional information below.
Uninstalling sipdroid fixed the dialpad

Any plan to release Sipdroid for Android 1.1 
because Cupcake still missing some Google application and we can not use it
for normal operation and the previous version of Sipdroid contain not
operate when i try it From www.hsc.com ,Thank's

I try to use with justvoip and impossible to make it work...
Know anyone the solution?

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1.
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?
limited version of sipdroid from market on hakuyro's adp1.5hr1

Please provide any additional information below.
tried registering with gizmo acct as well as pbxes.org acct, can somebody help?

The newest version is not even trying to register to the SIP server. 
Notification says "Registering..." and nothing happens. 

Confirmed by running tcpdump on the WLAN firewall.

Sipdroid fails to register to Asterisk Server. Obviously I have changed the
CLIENT-IP, DOMAIN.COM, and IP.CHANGED.FOR.SECURITY for security reasons. 

Sipdroid is giving up on passing authentication digest to Asterisk. 


REGISTER sip:domain.com SIP/2.0
Via: SIP/2.0/UDP 127.0.0.1:5060;rport;branch=z9hG4bK20481
Max-Forwards: 70
To: <sip:[email protected]>
From: <sip:[email protected]>;tag=z9hG4bK64544479
Call-ID: [email protected]
CSeq: 1 REGISTER
Contact: <sip:[email protected]>
Expires: 3600
User-Agent: mjsip stack 1.6
Content-Length: 0


SIP/2.0 100 Trying
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK20481;received=CLIENT-IP;rport=5060
From: <sip:[email protected]>;tag=z9hG4bK64544479
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:[email protected]>
Content-Length: 0



SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:5060;branch=z9hG4bK20481;received=CLIENT-IP;rport=5060
From: <sip:[email protected]>;tag=z9hG4bK64544479
To: <sip:[email protected]>;tag=as55f988eb
Call-ID: [email protected]
CSeq: 1 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5d25430a"
Content-Length: 0

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1.Receiving a Call.
2.
3.

What is the expected output? What do you see instead?
The phone was expected to ring when someone calls, but instead, it just
turn on the light, no sound or vibration.

What version of the product are you using? On what operating system?
Sipdroid-0.9.5-full.apk - g1/cupcake 1.5.1.

Please provide any additional information below.

What steps will reproduce the problem?
1. Running 5.0.2Hr5-A2SD - Haykuro
2. Configure for pbxes.org
3. Configure pbxes.org to point to desires sip settings

What is the expected output? What do you see instead?
Expected output would be to connect to pbxes.org as recommended.
I can get sipdroid to attempt registration if i set android to "only" use 
2G.  it will either fail with a 404 or timeout.  If I set android to 3G 
only I never see the orb stating the registration status.

What version of the product are you using? On what operating system?
Sipdroid-0.9-full.
5.0.2Hr5-A2SD - Haykuro

Please provide any additional information below.

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1. install cupcake JF ADP 1.5 ROM
2. Install siproid
3. 3g and EDGE options are greyed out for me (using cingular/atnt in US)
and siprdoid registration over wifi fails

What is the expected output? What do you see instead?

3g and EDGE options should NOT be greyed out and sipdroid registration
should not fail over WIFI too.

What version of the product are you using? On what operating system?
public beta version on android 1.5

Please provide any additional information below.

When I first installed sipdroid, before I even set it up, when I tried to
dial a contact, I would be prompted for which app to use (dialer, gv or
sipdroid).

After I set it up, I no longer get that prompt.

I went into the relevant apps and cleared the defaults but still no luck.

At first I had the market version but after that I installed the full version.

Any ideas?

Thanks

NOTE: This form is only for reporting bugs. For questions, comments, or
advice please visit:  http://groups.google.com/group/sipdroid-users

What steps will reproduce the problem?
1.adb install
2.
3.

What is the expected output? What do you see instead?


What version of the product are you using? On what operating system?
1.1  RC33

Please provide any additional information below.
I am getting failure [-12] when trying to install through adb    any ideas
also from phone browser, tried to download apk getting install unsuccessful