[asterisk-bugs] [Asterisk 0014220]: SIP INVITE packets are incorrectly truncated with 1.6.1 svn after approx 1020 characters

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=14220 
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Reported By:                riksta
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   14220
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           SVN 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6.1 
SVN Revision (number only!): 167727 
Request Review:              
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Date Submitted:             2009-01-12 12:02 CST
Last Modified:              2009-01-16 15:30 CST
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Summary:                    SIP INVITE packets are incorrectly truncated with
1.6.1 svn after approx 1020 characters
Description: 
Since updating from 1.6.1b4 to SVN-branch-1.6.1-r167727 i can no longer
make calls to my SIP trunk provider if my call has a long CALLERID(name)
string.

The sip trunk provider's techie says that my SIP INVITE packets are being
truncated... eg if i set the callerid name as a short string and the INVITE
packet was 1017 characters, the call goes through perfectly.

If i set the callerid name as a longer string and the INVITE packet was
1020 chars the call fails to be sent.

In #asterisk-dev Corydon76 explains that it is probably to do with the
fact that SIP INVITES are no longer are built in a static buffer (That was
part of the ast_str work) 


I will attach a sip debug log file of a successful and unsuccessful call
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---------------------------------------------------------------------- 
 (0098052) jamesgolovich (manager) - 2009-01-16 15:30
 http://bugs.digium.com/view.php?id=14220#c98052 
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I was poking around the code to see if anything jumped out at me and it
looks like theres another piece of unneeded code in add_sdp

        if (m_audio->len - m_audio->used < 2 || m_video->len -
m_video->used < 2 ||
                        m_text->len - m_text->used < 2 || a_text->len -
a_text->used < 2 ||
                        a_audio->len - a_audio->used < 2 || a_video->len -
a_video->used < 2)
                ast_log(LOG_WARNING, "SIP SDP may be truncated due to
undersized buffer!!\n"); 

Issue History 
Date Modified    Username       Field                    Change               
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2009-01-16 15:30 jamesgolovich  Note Added: 0098052                          
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