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Add an API call that initializes an RTP structure. We need this because chan_sip is cheeky and uses a temporary RTP structure for codec purposes, and the API calls that are used rely on the lock. (Pointed out on asterisk-dev by Andy Wang)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@48964 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2006-12-26 04:31:58 +00:00
parent 84a2b97b17
commit 9cc04e026d
3 changed files with 22 additions and 6 deletions
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24
main/rtp.c
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@@ -1798,6 +1798,23 @@ static struct ast_rtcp *ast_rtcp_new(void)
return rtcp;
}
/*!
* \brief Initialize a new RTP structure.
*
*/
void ast_rtp_new_init(struct ast_rtp *rtp)
{
ast_mutex_init(&rtp->bridge_lock);
rtp->them.sin_family = AF_INET;
rtp->us.sin_family = AF_INET;
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0xffff;
ast_set_flag(rtp, FLAG_HAS_DTMF);
return;
}
struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io_context *io, int rtcpenable, int callbackmode, struct in_addr addr)
{
struct ast_rtp *rtp;
@@ -1808,14 +1825,9 @@ struct ast_rtp *ast_rtp_new_with_bindaddr(struct sched_context *sched, struct io
if (!(rtp = ast_calloc(1, sizeof(*rtp))))
return NULL;
ast_mutex_init(&rtp->bridge_lock);
ast_rtp_new_init(rtp);
rtp->them.sin_family = AF_INET;
rtp->us.sin_family = AF_INET;
rtp->s = rtp_socket();
rtp->ssrc = ast_random();
rtp->seqno = ast_random() & 0xffff;
ast_set_flag(rtp, FLAG_HAS_DTMF);
if (rtp->s < 0) {
free(rtp);
ast_log(LOG_ERROR, "Unable to allocate socket: %s\n", strerror(errno));