diff options
author | jpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-06-06 16:39:20 +0000 |
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committer | jpeeler <jpeeler@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-06-06 16:39:20 +0000 |
commit | 8ed6031aad5c1b5f0d3cce3b427715829e45285f (patch) | |
tree | 63f870a38eefe1f503971d9139807da3ddb7cfa9 /channels/chan_sip.c | |
parent | eee2e227c9df2492521d7a4ed1ea0b987d88d1b8 (diff) |
Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables.
git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.4@120885 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels/chan_sip.c')
-rw-r--r-- | channels/chan_sip.c | 14 |
1 files changed, 11 insertions, 3 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index 50a7f4381..278528c80 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -1194,7 +1194,10 @@ static void temp_pvt_cleanup(void *); /*! \brief A per-thread temporary pvt structure */ AST_THREADSTORAGE_CUSTOM(ts_temp_pvt, temp_pvt_init, temp_pvt_cleanup); -AST_THREADSTORAGE(ast_rtp_buf, ast_rtp_buf_init); +static void ts_ast_rtp_destroy(void *); + +AST_THREADSTORAGE_CUSTOM(ts_audio_rtp, ts_audio_rtp_init, ts_ast_rtp_destroy); +AST_THREADSTORAGE_CUSTOM(ts_video_rtp, ts_video_rtp_init, ts_ast_rtp_destroy); /*! \todo Move the sip_auth list to AST_LIST */ static struct sip_auth *authl = NULL; /*!< Authentication list for realm authentication */ @@ -5063,7 +5066,7 @@ 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 */ #ifdef LOW_MEMORY - newaudiortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size()); + newaudiortp = ast_threadstorage_get(&ts_audio_rtp, ast_rtp_alloc_size()); #else newaudiortp = alloca(ast_rtp_alloc_size()); #endif @@ -5072,7 +5075,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) ast_rtp_pt_clear(newaudiortp); #ifdef LOW_MEMORY - newvideortp = ast_threadstorage_get(&ast_rtp_buf, ast_rtp_alloc_size()); + newvideortp = ast_threadstorage_get(&ts_video_rtp, ast_rtp_alloc_size()); #else newvideortp = alloca(ast_rtp_alloc_size()); #endif @@ -5607,6 +5610,11 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req) return 0; } +static void ts_ast_rtp_destroy(void *data) +{ + struct ast_rtp *tmp = data; + ast_rtp_destroy(tmp); +} /*! \brief Add header to SIP message */ static int add_header(struct sip_request *req, const char *var, const char *value) |