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Merge changes from team/group/sip-tcptls

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This set of changes introduces TCP and TLS support for chan_sip.  There are various
new options in configs/sip.conf.sample that are used to enable these features.  Also,
there is a document, doc/siptls.txt that describes some things in more detail.

This code was implemented by Brett Bryant and James Golovich.  It was reviewed
by Joshua Colp and myself.  A number of other people participated in the testing
of this code, but since it was done outside of the bug tracker, I do not have their
names.  If you were one of them, thanks a lot for the help!

(closes issue #4903, but with completely different code that what exists there.)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Russell Bryant
2008-01-18 22:04:33 +00:00
parent 1807acb9b0
commit b995c78c31
11 changed files with 1424 additions and 390 deletions
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16
configs/sip.conf.sample
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@@ -70,6 +70,16 @@ allowoverlap=no ; Disable overlap dialing support. (Default is yes)
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
; bindport is the local UDP port that Asterisk will listen on
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
tcpenable=yes ; Enable server for incoming TCP connections (default is yes)
tcpbindaddr=0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5062 (default is port 5060)
;tlsenable=no ; Enable server for incoming TLS (secure) connections (default is no)
;tlsbindaddr=0.0.0.0 ; IP address for TLS server to bind to (0.0.0.0) binds to all interfaces)
; Optionally add a port number, 192.168.1.1:5063 (default is port 5061)
;tlscertfile=asterisk.pem ; Certificate file (*.pem only) to use for TLS connections
; default is to look for "asterisk.pem" in current directory
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
@@ -320,7 +330,9 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;----------------------------------------- OUTBOUND SIP REGISTRATIONS ------------------------
; Asterisk can register as a SIP user agent to a SIP proxy (provider)
; Format for the register statement is:
; register => user[:secret[:authuser]]@host[:port][/extension]
; register => [transport://]user[:secret[:authuser]]@host[:port][/extension]
;
;
;
; If no extension is given, the 's' extension is used. The extension needs to
; be defined in extensions.conf to be able to accept calls from this SIP proxy
@@ -607,7 +619,7 @@ srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; User config options: Peer configuration:
; -------------------- -------------------
; context context
; callingpres callingpres
; callingpres callingpres
; permit permit
; deny deny
; secret secret