Added direct media support to ooh323 channel driver

options are documented in config sample
sample config rename to proper name - ooh323.conf

To change media address ooh323 send empty TCS if there was 
completed TCS exchange or send facility forwardedelements 
with new fast start proposal if not.
Then close transmit logical channels and renew TCS exchange.

If new fast start proposal is received then ooh323 stack call back
channel driver routine to change rtp address in the rtp instance.
If empty TCS is received then close transmit logical channels and
renew TCS exchange

Review: https://reviewboard.asterisk.org/r/1607/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369613 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 204 insertions, 0 deletions

diff --git a/configs/ooh323.conf.sample b/configs/ooh323.conf.sample
new file mode 100644
index 000000000..d8ccdcf00
--- /dev/null
+++ b/configs/ooh323.conf.sample
@@ -0,0 +1,204 @@
+; ---------------------------------------------------------------------------------
+; ---  ******* IMPORTANT NOTE ***********
+; ---
+; ---  This module is currently unsupported.  Use it at your own risk.
+; ---
+; ---------------------------------------------------------------------------------
+
+; Objective System's H323 Configuration example for Asterisk
+; ooh323c driver configuration
+;
+; [general] section defines global parameters
+;
+; This is followed by profiles which can be of three types - user/peer/friend
+; Name of the user profile should match with the h323id of the user device.
+; For peer/friend profiles, host ip address must be provided as "dynamic" is
+; not supported as of now.
+;
+; Syntax for specifying a H323 device in extensions.conf is
+; For Registered peers/friends profiles:
+;        OOH323/name where name is the name of the peer/friend profile.
+;
+; For unregistered H.323 phones:
+;        OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
+;                          alias
+;
+; For dialing into another asterisk peer at a specific exten
+;       OOH323/exten/peer OR OOH323/exten@ip
+;
+; Domain name resolution is not yet supported.
+;
+; When a H.323 user calls into asterisk, his H323ID is matched with the profile
+; name and context is determined to route the call
+;
+; The channel driver will register all global aliases and aliases defined in
+; peer profiles with the gatekeeper, if one exists. So, that when someone
+; outside our pbx (non-user) calls an extension, gatekeeper will route that
+; call to our asterisk box, from where it will be routed as per dial plan.
+
+
+[general]
+;Define the asetrisk server h323 endpoint
+
+;The port asterisk should listen for incoming H323 connections.
+;Default - 1720
+;port=1720
+
+;The dotted IP address asterisk should listen on for incoming H323
+;connections
+;Default - tries to find out local ip address on it's own
+bindaddr=0.0.0.0
+
+;This parameter indicates whether channel driver should register with
+;gatekeeper as a gateway or an endpoint.
+;Default - no
+;gateway=no
+
+;Whether asterisk should use fast-start and tunneling for H323 connections.
+;Default - yes
+;faststart=no
+;h245tunneling=no
+
+;Whether media wait for connect
+;Default - No
+;mediawaitforconnect=yes
+
+;H323-ID to be used for asterisk server
+;Default - Asterisk PBX
+h323id=ObjSysAsterisk
+e164=100
+
+;CallerID to use for calls
+;Default - Same as h323id
+callerid=asterisk
+
+;Whether this asterisk server will use gatekeeper.
+;Default - DISABLE
+;gatekeeper = DISCOVER
+;gatekeeper = a.b.c.d
+gatekeeper = DISABLE
+
+;Location for H323 log file
+;Default - /var/log/asterisk/h323_log
+;logfile=/var/log/asterisk/h323_log
+
+
+;Following values apply to all users/peers/friends defined below, unless
+;overridden within their client definition
+
+;Sets default context all clients will be placed in.
+;Default - default
+context=default
+
+;Sets rtptimeout for all clients, unless overridden
+;Default - 60 seconds
+;rtptimeout=60	    ; Terminate call if 60 seconds of no RTP activity
+                    ; when we're not on hold
+
+;Type of Service
+;Default - none (lowdelay, thoughput, reliability, mincost, none)
+;tos=lowdelay
+
+;amaflags = default
+
+;The account code used by default for all clients.
+;accountcode=h3230101
+
+;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
+;Default - ulaw
+; ONLY ulaw, gsm, g729 and g7231 supported as of now
+disallow=all     ;Note order of disallow/allow is important.
+allow=gsm
+allow=ulaw
+
+
+; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
+; h245alphanumeric, h245signal.
+;Default - rfc 2833
+dtmfmode=rfc2833
+;
+; round trip delay request, default = 0,0 (not send)
+; x - count of unreplied requests before hangup, y - interval in sec between requests
+;
+;roundtrip=x,y
+
+;
+; FAX detection will cause the OOH323 channel to jump to the 'fax' extension (if it exists)
+; based one or more events being detected. The events that can be detected are an incoming
+; CNG tone or an incoming T.38 RequestMode packet
+;
+; yes - enable both detection (CNG & T.38)
+; no - disable both
+; cng - enable CNG detection (default)
+; t38 - enable T.38 request detection
+;
+faxdetect = cng
+
+; User/peer/friend definitions:
+; User config options                    Peer config options
+; ------------------                     -------------------
+; context
+; disallow                               disallow
+; allow                                  allow
+; accountcode                            accountcode
+; amaflags                               amaflags
+; dtmfmode                               dtmfmode
+; rtptimeout                             ip
+;                                        port
+;                                        h323id
+;                                        email
+;                                        url
+;                                        e164
+;                                        rtptimeout
+
+;
+
+;
+;	direct rtp between two remote peers, disabled by default
+;	can be specified globally or per peer/user section
+;
+directmedia=no
+;
+;	early direct rtp (try to establish direct rtp before asnwer)
+;	disabled by default, auto enabled by directmedia is enabled
+;	can be disabled escpecially if directmedia is enabled.
+;	can be specified globally or per peer/user section
+;
+;
+directrtpsetup=no
+
+;Define users here
+;Section header is extension
+[myuser1]
+type=user
+context=context1
+disallow=all
+allow=gsm
+allow=ulaw
+
+
+
+[mypeer1]
+type=peer
+context=context2
+ip=a.b.c.d   ; UPDATE with appropriate ip address
+port=1720    ; UPDATE with appropriate port
+e164=101
+
+
+
+[myfriend1]
+type=friend
+context=default
+ip=10.0.0.82   ; UPDATE with appropriate ip address
+port=1820    ; UPDATE with appropriate port
+disallow=all
+allow=ulaw
+e164=12345
+rtptimeout=60
+dtmfmode=rfc2833
+
+
+
+
+