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Add SIP/RTP video support, video enable app_echo, start on RTCP

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1128 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Mark Spencer
2003-06-28 16:40:02 +00:00
parent 71c9cbb2b1
commit f5e13431a5
11 changed files with 589 additions and 173 deletions
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20
channel.c
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@@ -1,4 +1,4 @@
/*
/*
* Asterisk -- A telephony toolkit for Linux.
*
* Channel Management
@@ -1269,6 +1269,17 @@ int ast_prod(struct ast_channel *chan)
return 0;
}
int ast_write_video(struct ast_channel *chan, struct ast_frame *fr)
{
int res;
if (!chan->pvt->write_video)
return 0;
res = ast_write(chan, fr);
if (!res)
res = 1;
return res;
}
int ast_write(struct ast_channel *chan, struct ast_frame *fr)
{
int res = -1;
@@ -1306,6 +1317,13 @@ int ast_write(struct ast_channel *chan, struct ast_frame *fr)
if (chan->pvt->send_text)
res = chan->pvt->send_text(chan, (char *) fr->data);
break;
case AST_FRAME_VIDEO:
/* XXX Handle translation of video codecs one day XXX */
if (chan->pvt->write_video)
res = chan->pvt->write_video(chan, fr);
else
res = 0;
break;
default:
if (chan->pvt->write) {
if (chan->pvt->writetrans) {