diff options
| author | Mark Michelson <mmichelson@digium.com> | 2015-01-29 20:54:46 +0000 |
|---|---|---|
| committer | Mark Michelson <mmichelson@digium.com> | 2015-01-29 20:54:46 +0000 |
| commit | fe76d4829fa0cc74c89dac1caab19f1fb4332acf (patch) | |
| tree | fa1c16b371c0663c4eae20ee7e05ab508a58e01f /res/res_pjsip | |
| parent | 8357ffab9c939062a2a6d546a90c766f519d30ca (diff) | |
Use SIPS URIs in Contact headers when appropriate.
RFC 3261 sections 8.1.1.8 and 12.1.1 dictate specific
scenarios when we are required to use SIPS URIs in Contact
headers. Asterisk's non-compliance with this could actually
cause calls to get dropped when communicating with clients
that are strict about checking the Contact header.
Both of the SIP stacks in Asterisk suffered from this issue.
This changeset corrects the behavior in chan_sip.
ASTERISK-24646 #close
Reported by Stephan Eisvogel
Review: https://reviewboard.asterisk.org/r/4346
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Merged revisions 431423 from http://svn.asterisk.org/svn/asterisk/branches/11
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Merged revisions 431424 from http://svn.asterisk.org/svn/asterisk/branches/13
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'res/res_pjsip')
0 files changed, 0 insertions, 0 deletions