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use the module pool for the RTP sessions instead of the session pool

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git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@17111 d0543943-73ff-0310-b7d9-9358b9ac24b2
This commit is contained in:
Moises Silva 2010-03-26 16:20:51 +00:00
parent 0fd16cc446
commit 8282194564
1 changed files with 8 additions and 2 deletions
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10
src/mod/codecs/mod_sangoma_codec/mod_sangoma_codec.c
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@ -72,6 +72,9 @@ unsigned long long g_next_session_id = 0;
/* hash of sessions (I think a linked list suits better here, but FS does not have the data type) */
switch_hash_t *g_sessions_hash = NULL;
/* global memory pool provided by FS */
switch_memory_pool_t *g_pool = NULL;
typedef struct vocallo_codec_s {
int codec_id; /* vocallo codec ID */
int iana; /* IANA code to register in FS */
@ -196,13 +199,14 @@ static int sangoma_create_rtp(void *usr_priv, sngtc_codec_request_leg_t *codec_r
local_ip, rtp_port, codec_ip, codec_reply_leg->codec_udp_port, iana,
codec_reg_leg->sample_rate, codec_reg_leg->ms*1000, sess->sessid);
/* create the RTP socket */
/* create the RTP socket, dont use the session pool since the session may go away while the RTP socket should linger around
* until sangoma_transcode decides to kill it */
rtp_session = switch_rtp_new(local_ip, rtp_port,
codec_ip, codec_reply_leg->codec_udp_port,
iana,
codec_reg_leg->ms*8, /* samples per interval, FIXME: do based on sampling rate */
codec_reg_leg->ms*1000, /* ms per packet */
flags, NULL, &err, sess->pool);
flags, NULL, &err, g_pool);
if (!rtp_session) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed to create switch rtp session: %s\n", err);
@ -900,6 +904,8 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_sangoma_codec_load)
return SWITCH_STATUS_FALSE;
}
g_pool = pool;
g_init_cfg.log = sangoma_logger;
g_init_cfg.create_rtp = sangoma_create_rtp;
g_init_cfg.destroy_rtp = sangoma_destroy_rtp;