Files · 9953f4f40eddceccd4d99f07ffe9e8a380e371e9 · Voice / asterisk · GitLab

Snippets Groups Projects

Make SIP early media work more efficiently without so many reinvites

Mark Spencer authored 18 years ago

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

9953f4f4

9953f4f4 18 years ago

Name	Last commit	Last update
agi
apps
build_tools
cdr
channels
codecs
configs
contrib
cygwin
db1-ast
doc
editline
formats
funcs
images
include
keys
mxml
pbx
redhat
res
sounds
static-http
stdtime
utils
.cleancount
BUGS
CHANGES
COPYING
CREDITS
LICENSE
Makefile
README
UPGRADE.txt
acinclude.m4
acl.c
aclocal.m4
aescrypt.c
aeskey.c
aesopt.h
aestab.c
alaw.c
app.c
ast_expr2.c
ast_expr2.fl
ast_expr2.h
ast_expr2.y
ast_expr2f.c
asterisk.8
asterisk.c
asterisk.sgml
astmm.c
autoservice.c
bootstrap.sh
buildinfo.c
callerid.c
cdr.c
channel.c
chanvars.c
cli.c
coef_in.h
coef_out.h
config.c
config.guess
config.sub
configure
configure.ac
cryptostub.c
db.c
devicestate.c
dlfcn.c
dns.c
dnsmgr.c
dsp.c
ecdisa.h
enum.c
file.c
frame.c
fskmodem.c
http.c
image.c
indications.c
install-sh
io.c
jitterbuf.c
jitterbuf.h
loader.c
logger.c
makeopts.in
manager.c
md5.c
missing
mkinstalldirs
muted.c
netsock.c
pbx.c
plc.c
poll.c
privacy.c
rtp.c