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Merged revisions 48964 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r48964 | file | 2006-12-25 23:31:58 -0500 (Mon, 25 Dec 2006) | 2 lines

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/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp 2006-12-26 04:34:07 +00:00
parent b3ab530077
commit 7f61b822c1
3 changed files with 21 additions and 6 deletions
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2
channels/chan_sip.c
View File

@ -4800,10 +4800,12 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Initialize the temporary RTP structures we use to evaluate the offer from the peer */
newaudiortp = alloca(ast_rtp_alloc_size());
memset(newaudiortp, 0, ast_rtp_alloc_size());
ast_rtp_new_init(newaudiortp);
ast_rtp_pt_clear(newaudiortp);
newvideortp = alloca(ast_rtp_alloc_size());
memset(newvideortp, 0, ast_rtp_alloc_size());
ast_rtp_new_init(newvideortp);
ast_rtp_pt_clear(newvideortp);
/* Update our last rtprx when we receive an SDP, too */

1
include/asterisk/rtp.h
View File

@ -223,6 +223,7 @@ int ast_rtcp_send_h261fur(void *data);
char *ast_rtp_get_quality(struct ast_rtp *rtp); /*! \brief Return RTCP quality string */
void ast_rtp_init(void); /*! Initialize RTP subsystem */
int ast_rtp_reload(void); /*! reload rtp configuration */
void ast_rtp_new_init(struct ast_rtp *rtp);
/*! Set codec preference */
int ast_rtp_codec_setpref(struct ast_rtp *rtp, struct ast_codec_pref *prefs);

24
main/rtp.c
View File

@ -1860,6 +1860,23 @@ 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;
@ -1870,14 +1887,9 @@ 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));