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)

........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48965 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index e21e9670a3e772f62c1a7b57b36bc4d2cc68756d..b02d0a7e1e4ebbb01b80bb24a5d45e18b8e35447 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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 */

diff --git a/include/asterisk/rtp.h b/include/asterisk/rtp.h
index a93f39261f7a220c8cba895a84052ac4466b63d1..2edff8cadbcdc8437659ce8d7121cfd005c152aa 100644 (file)
--- a/include/asterisk/rtp.h
+++ b/include/asterisk/rtp.h
@@ -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);

diff --git a/main/rtp.c b/main/rtp.c
index 88ad606da2e22c23b0f512047bd0b781aa6b8c13..b3fbf684a57f41f6d8426bdb284b9907c57e1c16 100644 (file)
--- a/main/rtp.c
+++ b/main/rtp.c
@@ -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));