summaryrefslogtreecommitdiff
path: root/doc/rtp-packetization.txt
diff options
context:
space:
mode:
authorRussell Bryant <russell@russellbryant.com>2010-11-11 22:14:25 +0000
committerRussell Bryant <russell@russellbryant.com>2010-11-11 22:14:25 +0000
commit893ca656af419e58c8dd675274d4a4d59b22cc03 (patch)
tree8b9307baeee40cb5429b1fada5b3da28ec15b536 /doc/rtp-packetization.txt
parent99a698efb7c0bc8548c032b37692da8ec13be9ea (diff)
Merged revisions 294740 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r294740 | russell | 2010-11-11 16:13:38 -0600 (Thu, 11 Nov 2010) | 11 lines Remove most of the contents of the doc dir in favor of the wiki content. This merge does the following things: * Removes most of the contents from the doc/ directory in favor of the wiki - http://wiki.asterisk.org/ * Updates the build_tools/prep_tarball script to know how to export the contents of the wiki in both PDF and plain text formats so that the documentation is still included in Asterisk release tarballs. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294741 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'doc/rtp-packetization.txt')
-rw-r--r--doc/rtp-packetization.txt75
1 files changed, 0 insertions, 75 deletions
diff --git a/doc/rtp-packetization.txt b/doc/rtp-packetization.txt
deleted file mode 100644
index c558a538e..000000000
--- a/doc/rtp-packetization.txt
+++ /dev/null
@@ -1,75 +0,0 @@
-Overview
--------
-Asterisk currently supports configurable RTP packetization per codec for
-select RTP-based channels.
-
-Channels
--------
-These channel drivers allow RTP packetization on a user/peer/friend
-or global level:
- chan_sip
- chan_skinny
- chan_h323
- chan_ooh323 (Asterisk-Addons)
- chan_gtalk
- chan_jingle
-
-Configuration
--------
-To set a desired packetization interval on a specific codec,
-append that inteval to the allow= statement.
-
-Example:
-allow=ulaw:30,alaw,g729:60
-
-No packetization is specified in the case of alaw in this example,
-so the default of 20ms is used.
-
-Autoframing
--------
-In addition, chan_sip has the ability to negotiate the desired
-framing at call establishment.
-
-In sip.conf if autoframing=yes is set in the global section, then
-all calls will try to set the packetization based on the remote
-endpoint's preferences. This behaviour depends on the endpoints
-ability to present the desired packetization (ptime:) in the SDP.
-If the endpoint does not include a ptime attribute, the call will
-be established with 20ms packetization.
-
-Autoframing can be set at the global level or on a user/peer/friend
-basis. If it is enabled at the global level, it applies to all
-users/peers/friends regardless of their prefered codec packetization.
-
-Codec framing options
--------
-The following table lists the minimum and maximum values that are
-valid per codec, as well as the increment value used for each.
-Please note that the maximum values here are only recommended
-maximums, and should not exceed the RTP MTU.
-
-Name Min Max Default Increment
-g723 30 300 30 30
-gsm 20 300 20 20
-ulaw 10 150 20 10
-alaw 10 150 20 10
-g726 10 300 20 10
-ADPCM 10 300 20 10
-SLIN 10 70 20 10
-lpc10 20 20 20 20
-g729 10 230 20 10
-speex 10 60 20 10
-ilbc 30 30 30 30
-g726_aal2 10 300 20 10
-
-Invalid framing options are handled based on the following rules:
- 1. If the specified framing is less than the codec's minimum, then
- the minimum value is used.
- 2. If the specific framing is greater than the codec's maximum, then
- the maximum value is used
- 3. If the specificed framing does not meet the increment requirement,
- the specified framing is rounded down to the closest valid
- framing options.
- example allow=ulaw:33 will set the codec to 30ms framing
- 4. If no framing is specified in the allow= directive, then the
- codec default is used.