[Asterisk-code-review] res pjsip sdp rtp: No rtpmap for static RTP payload IDs in SDP. (asterisk[master])
Alexander Traud
asteriskteam at digium.com
Thu Apr 13 04:07:18 CDT 2017
Hello Anonymous Coward #1000019, Sean Bright, Joshua Colp,
I'd like you to reexamine a change. Please visit
https://gerrit.asterisk.org/5431
to look at the new patch set (#2).
Change subject: res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.
......................................................................
res_pjsip_sdp_rtp: No rtpmap for static RTP payload IDs in SDP.
This saves around 100 bytes when G.711, G.722, G.729, and GSM are advertised in
SDP. This reduces the chance to hit the MTU bearer of 1300 bytes for SIP over
UDP, if many codecs are allowed in Asterisk. This new feature is enabled
together with the optional feature compact_headers=yes via the file pjsip.conf.
ASTERISK-26932 #close
Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
---
M channels/chan_sip.c
M include/asterisk/rtp_engine.h
M main/rtp_engine.c
M res/res_pjsip_sdp_rtp.c
4 files changed, 26 insertions(+), 15 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/31/5431/2
--
To view, visit https://gerrit.asterisk.org/5431
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Gerrit-MessageType: newpatchset
Gerrit-Change-Id: Iaa556ab4c8325cd34c334387ab2847fab07b1689
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Alexander Traud <pabstraud at compuserve.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Sean Bright <sean.bright at gmail.com>
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