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udptl: Dead code elimination. ast_udptl_bridge was not used.

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Removing dead code starting with ast_udptl_bridge() eliminated the code in
this change.

Note: This code has actually been dead since Asterisk v1.4 when it was
first put in.

Review: https://reviewboard.asterisk.org/r/3079/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/12@404354 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2013-12-19 17:03:55 +00:00
parent 2a1275abf9
commit c8b02bc487
4 changed files with 0 additions and 299 deletions
Download Patch File Download Diff File Expand all files Collapse all files

76
channels/chan_sip.c
View File

@@ -1535,8 +1535,6 @@ static int process_crypto(struct sip_pvt *p, struct ast_rtp_instance *rtp, struc
/*------ T38 Support --------- */
static int transmit_response_with_t38_sdp(struct sip_pvt *p, char *msg, struct sip_request *req, int retrans);
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan);
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl);
static void change_t38_state(struct sip_pvt *p, int state);
/*------ Session-Timers functions --------- */
@@ -3571,13 +3569,6 @@ static struct sip_registry *registry_addref(struct sip_registry *reg, char *tag)
return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}
/*! \brief Interface structure with callbacks used to connect to UDPTL module*/
static struct ast_udptl_protocol sip_udptl = {
.type = "SIP",
.get_udptl_info = sip_get_udptl_peer,
.set_udptl_peer = sip_set_udptl_peer,
};
static void append_history_full(struct sip_pvt *p, const char *fmt, ...)
__attribute__((format(printf, 2, 3)));
@@ -32457,67 +32448,6 @@ static int reload_config(enum channelreloadreason reason)
return 0;
}
static struct ast_udptl *sip_get_udptl_peer(struct ast_channel *chan)
{
struct sip_pvt *p;
struct ast_udptl *udptl = NULL;
p = ast_channel_tech_pvt(chan);
if (!p) {
return NULL;
}
sip_pvt_lock(p);
if (p->udptl && ast_test_flag(&p->flags[0], SIP_DIRECT_MEDIA)) {
udptl = p->udptl;
}
sip_pvt_unlock(p);
return udptl;
}
static int sip_set_udptl_peer(struct ast_channel *chan, struct ast_udptl *udptl)
{
struct sip_pvt *p;
/* Lock the channel and the private safely. */
ast_channel_lock(chan);
p = ast_channel_tech_pvt(chan);
if (!p) {
ast_channel_unlock(chan);
return -1;
}
sip_pvt_lock(p);
if (p->owner != chan) {
/* I suppose it could be argued that if this happens it is a bug. */
ast_debug(1, "The private is not owned by channel %s anymore.\n", ast_channel_name(chan));
sip_pvt_unlock(p);
ast_channel_unlock(chan);
return 0;
}
if (udptl) {
ast_udptl_get_peer(udptl, &p->udptlredirip);
} else {
memset(&p->udptlredirip, 0, sizeof(p->udptlredirip));
}
if (!ast_test_flag(&p->flags[0], SIP_GOTREFER)) {
if (!p->pendinginvite) {
ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s\n",
p->callid, ast_sockaddr_stringify(udptl ? &p->udptlredirip : &p->ourip));
transmit_reinvite_with_sdp(p, TRUE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
ast_debug(3, "Deferring reinvite on SIP '%s' - It's UDPTL will be redirected to IP %s\n",
p->callid, ast_sockaddr_stringify(udptl ? &p->udptlredirip : &p->ourip));
ast_set_flag(&p->flags[0], SIP_NEEDREINVITE);
}
}
/* Reset lastrtprx timer */
p->lastrtprx = p->lastrtptx = time(NULL);
sip_pvt_unlock(p);
ast_channel_unlock(chan);
return 0;
}
static int sip_allow_anyrtp_remote(struct ast_channel *chan1, struct ast_rtp_instance *instance, const char *rtptype)
{
struct sip_pvt *p;
@@ -34511,9 +34441,6 @@ static int load_module(void)
/* Register all CLI functions for SIP */
ast_cli_register_multiple(cli_sip, ARRAY_LEN(cli_sip));
/* Tell the UDPTL subdriver that we're here */
ast_udptl_proto_register(&sip_udptl);
/* Tell the RTP engine about our RTP glue */
ast_rtp_glue_register(&sip_rtp_glue);
@@ -34644,9 +34571,6 @@ static int unload_module(void)
/* Unregister CLI commands */
ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));
/* Disconnect from UDPTL */
ast_udptl_proto_unregister(&sip_udptl);
/* Disconnect from RTP engine */
ast_rtp_glue_unregister(&sip_rtp_glue);