From 88efcea05e9e81aadf916d549dbcceeadf0387f3 Mon Sep 17 00:00:00 2001 From: "Kevin P. Fleming" Date: Wed, 25 Oct 2006 00:32:23 +0000 Subject: Merged revisions 46154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r46154 | kpfleming | 2006-10-24 19:26:17 -0500 (Tue, 24 Oct 2006) | 2 lines add passthrough and file format support for G.722 16KHz audio (issue #5084, original patch by andrew, updated by mithraen) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@46155 65c4cc65-6c06-0410-ace0-fbb531ad65f3 --- main/rtp.c | 2 ++ 1 file changed, 2 insertions(+) (limited to 'main/rtp.c') diff --git a/main/rtp.c b/main/rtp.c index 589584dee..c34f56b11 100644 --- a/main/rtp.c +++ b/main/rtp.c @@ -1330,6 +1330,7 @@ static struct { {{1, AST_FORMAT_G729A}, "audio", "G729"}, {{1, AST_FORMAT_SPEEX}, "audio", "speex"}, {{1, AST_FORMAT_ILBC}, "audio", "iLBC"}, + {{1, AST_FORMAT_G722}, "audio", "G722"}, {{1, AST_FORMAT_G726_AAL2}, "audio", "AAL2-G726-32"}, {{0, AST_RTP_DTMF}, "audio", "telephone-event"}, {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"}, @@ -1356,6 +1357,7 @@ static struct rtpPayloadType static_RTP_PT[MAX_RTP_PT] = { [6] = {1, AST_FORMAT_ADPCM}, /* 16 kHz */ [7] = {1, AST_FORMAT_LPC10}, [8] = {1, AST_FORMAT_ALAW}, + [9] = {1, AST_FORMAT_G722}, [10] = {1, AST_FORMAT_SLINEAR}, /* 2 channels */ [11] = {1, AST_FORMAT_SLINEAR}, /* 1 channel */ [13] = {0, AST_RTP_CN}, -- cgit v1.2.3