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Fix G.729 name annoucement to Open H.323 and inform Asterisk's RTP stack specifically which codec we are expecting.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@4508 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Jeremy McNamara
2004-12-21 00:07:56 +00:00
parent 9f6e3aed8a
commit f4d1002166
3 changed files with 36 additions and 14 deletions
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26
channels/chan_h323.c
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@@ -1166,15 +1166,25 @@ struct rtp_info *external_rtp_create(unsigned call_reference, const char * token
int progress(unsigned call_reference, const char *token, int inband);
/**
* Definition taken from rtp.c for rtpPayloadType because we need it here.
*/
struct rtpPayloadType {
int isAstFormat; /* whether the following code is an AST_FORMAT */
int code;
};
/**
* Call-back function passing remote ip/port information from H.323 to asterisk
*
* Returns nothing
*/
void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort, const char *token)
void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int remotePort, const char *token, int pt)
{
struct oh323_pvt *pvt = NULL;
struct sockaddr_in them;
struct rtpPayloadType rtptype;
int fmt;
/* Find the call or allocate a private structure if call not found */
pvt = find_call(call_reference, token);
@@ -1182,6 +1192,19 @@ void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int rem
ast_log(LOG_ERROR, "Something is wrong: rtp\n");
return;
}
rtptype = ast_rtp_lookup_pt(pvt->rtp, pt);
fmt = ast_best_codec(pvt->owner->nativeformats);
ast_mutex_lock(&pvt->lock);
pvt->owner->nativeformats = rtptype.code;
pvt->owner->readformat = fmt;
pvt->owner->writeformat = fmt;
ast_set_read_format(pvt->owner, pvt->owner->readformat);
ast_set_write_format(pvt->owner, pvt->owner->writeformat);
ast_mutex_unlock(&pvt->lock);
them.sin_family = AF_INET;
/* only works for IPv4 */
them.sin_addr.s_addr = inet_addr(remoteIp);
@@ -1191,6 +1214,7 @@ void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int rem
if (pvt->options.progress_audio) {
progress(call_reference, token, 1);
}
return;
}