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Don't do reinvite if both parties talk diffrent codecs

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1752 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Martin Pycko
2003-11-15 00:52:49 +00:00
parent 6f9da41a6c
commit e3b0a66805
2 changed files with 8 additions and 0 deletions
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7
channels/chan_sip.c
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@@ -6287,10 +6287,17 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
return 0;
}
static int sip_get_codec(struct ast_channel *chan)
{
struct sip_pvt *p = chan->pvt->pvt;
return p->capability;
}
static struct ast_rtp_protocol sip_rtp = {
get_rtp_info: sip_get_rtp_peer,
get_vrtp_info: sip_get_vrtp_peer,
set_rtp_peer: sip_set_rtp_peer,
get_codec: sip_get_codec,
};
int load_module()

1
include/asterisk/rtp.h
View File

@@ -39,6 +39,7 @@ struct ast_rtp_protocol {
struct ast_rtp *(*get_rtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */
struct ast_rtp *(*get_vrtp_info)(struct ast_channel *chan); /* Get RTP struct, or NULL if unwilling to transfer */
int (*set_rtp_peer)(struct ast_channel *chan, struct ast_rtp *peer, struct ast_rtp *vpeer); /* Set RTP peer */
int (*get_codec)(struct ast_channel *chan);
char *type;
struct ast_rtp_protocol *next;
};