Skip to content
Snippets Groups Projects
Select Git revision
  • 14666-no-g722
  • asterisk_rdkb
  • dev_fxs_no_wb
  • 14600-wenpeng
  • fix_fallback
  • 14666_fxs_no_wideband_codec
  • fix_srv_records
  • wenpeng-registration-test
  • fix_deadlock_in_bridge_peer_functions
  • devel default protected
  • securesip_pjproject_part1
  • PJ_Example_Review
  • fix_internal_calls_auth
  • fix_internal_calls
  • Secure_SIPtelephony_part1
  • SIPtelephony_DECT_syslog
  • syslog_develop_phase1
  • merge-asterisk-20.8.1
  • asterisk_rdkb_temp
  • bug_14682_voicemail
  • 22.0.0-pre1
  • 21.4.2
  • 20.9.2
  • 18.24.2
  • certified-20.7-cert2
  • certified-18.9-cert11
  • 21.4.1
  • 20.9.1
  • 18.24.1
  • 21.4.0
  • 20.9.0
  • 18.24.0
  • certified-20.7-cert1
  • certified-18.9-cert10
  • 21.4.0-rc1
  • 20.9.0-rc1
  • 18.24.0-rc1
  • 21.3.1
  • 20.8.1
  • 18.23.1
40 results
Compare
user avatar
add a bit of documentation on what the stun code in rtp.c does
Luigi Rizzo authored 
(which is very little, at the moment).

Eventually, when the functionality is extended, the changes can be merged
back to 1.4. At the moment this is pointless.

Note, this change is whitespace only.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74571 65c4cc65-6c06-0410-ace0-fbb531ad65f3
deb98f98
History
user avatar deb98f98
History
Name Last commit Last update