Use SIPS URIs in Contact headers when appropriate.
authorMark Michelson <mmichelson@digium.com>
Thu, 29 Jan 2015 21:02:23 +0000 (21:02 +0000)
committerMark Michelson <mmichelson@digium.com>
Thu, 29 Jan 2015 21:02:23 +0000 (21:02 +0000)
commit034798e37e0a7471d2f213ef7b21157b7714e293
treec748fcab9bb220a307494f0b0b647c6c49fad5f4
parentfe76d4829fa0cc74c89dac1caab19f1fb4332acf
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 res_pjsip/chan_pjsip.c

Review: https://reviewboard.asterisk.org/r/4345
........

Merged revisions 431426 from http://svn.asterisk.org/svn/asterisk/branches/13

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
res/res_pjsip.c
res/res_pjsip_sips_contact.c [new file with mode: 0644]