pyj4104 / sipdroid

Automatically exported from code.google.com/p/sipdroid
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SIP over TCP not working properly. Calls get dropped after 30 seconds. #311

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago
NOTE: This form is only for reporting bugs. For problems, questions, or
comments, please visit:  http://groups.google.com/group/sipdroid-users

*************************************************
* Read the FAQ                                  *
*    Check if the bug has already been reported *
*        No doubles please                      *
*************************************************

Did you carefully read above and decide this means of communication is the
right for your notice? Don't just list an incompatible SIP server or
device, please. Try to look behind the scenes if you like to
contribute here. Thanks!

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What steps will reproduce the problem?
1. Place a call to a FreeSWITCH server over TCP
2. Wait 30 seconds
3. FreeSWITCH server sends a BYE and drops the call

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

Call should not drop after 30 seconds.

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

1.3.10. Android 1.5 (tried 1.6 with same results)

Which SIP server are you using? What happens with PBXes?

FreeSWITCH (latest SVN)which is based on a standards compliant SIP stack 
(Sofia by Nokia).

Which type of network are you using?

gigabit for server, 802.11g for android phone behind same network.

Please provide any additional information below.

This url includes a siptrace of the call and graph. 
http://pastebin.freeswitch.org/12064

One of the FreeSWITCH developers believes the issue is that sipdroid is not 
specifying "transport=TCP" in the contact field. This causes the server to 
fall back to UDP which the client is not listening on. After 30 seconds the 
call gets dropped.

Original issue reported on code.google.com by carlos.t...@gmail.com on 5 Feb 2010 at 9:58

Attachments:

GoogleCodeExporter commented 9 years ago
Figured this one out on my own. The class SipdroidEngine builds the contact 
header 
for the client. It never allows for the suffix "transport=" to be added. The 
simple 
addition below resolves this. Without this suffix FreeSWITCH assumes the client 
is 
talking over UDP and eventually drops the call. I believe this field is a 
requirement 
as per the SIP RFCS when communicating over TCP.

Can this be included in the next release? Does it break interoperability with 
pbxes?

Original comment by carlos.t...@gmail.com on 8 Feb 2010 at 8:31

Attachments:

GoogleCodeExporter commented 9 years ago

Original comment by pmerl...@googlemail.com on 9 Feb 2010 at 11:46

GoogleCodeExporter commented 9 years ago

Original comment by pmerl...@googlemail.com on 12 Feb 2010 at 1:41

GoogleCodeExporter commented 9 years ago
Not really sure if this issue can be reopened, but these symptoms are exactly 
the same as what I am seeing on 2.0.1 and 2.2pure and a 3G connection. 
Everything works great over TCP until 30 seconds into a call and the call is 
dropped. 

I've tried checking for the posted patch in the code of the latest version, but 
it appears as if it has either been removed or replaced by other objects, but I 
cannot locate the same functionality.

Can someone please check this?

Original comment by Qui...@gmail.com on 29 May 2011 at 1:17

GoogleCodeExporter commented 9 years ago
We are having the same problem on Obihai ata devices. Calls drop after 30 sec. 
Any thoughts?

Thank you
Robert

Original comment by mehr...@gmail.com on 20 Feb 2015 at 1:24