[svn-commits] oej: trunk r89609 - /trunk/channels/chan_sip.c
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Mon Nov 26 14:55:09 CST 2007
Author: oej
Date: Mon Nov 26 14:55:09 2007
New Revision: 89609
URL: http://svn.digium.com/view/asterisk?view=rev&rev=89609
Log:
Formatting changes, cleaning up some code
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=89609&r1=89608&r2=89609
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Nov 26 14:55:09 2007
@@ -1492,6 +1492,15 @@
static struct ast_config *notify_types; /*!< The list of manual NOTIFY types we know how to send */
+/*! some list management macros. */
+
+#define UNLINK(element, head, prev) do { \
+ if (prev) \
+ (prev)->next = (element)->next; \
+ else \
+ (head) = (element)->next; \
+ } while (0)
+
/*---------------------------- Forward declarations of functions in chan_sip.c */
/*! \note This is added to help splitting up chan_sip.c into several files
in coming releases */
@@ -1886,14 +1895,6 @@
return errorvalue;
}
-/**--- some list management macros. **/
-
-#define UNLINK(element, head, prev) do { \
- if (prev) \
- (prev)->next = (element)->next; \
- else \
- (head) = (element)->next; \
- } while (0)
/*! \brief Interface structure with callbacks used to connect to RTP module */
static struct ast_rtp_protocol sip_rtp = {
@@ -2954,6 +2955,7 @@
}
}
+/*! Destroy mailbox subscriptions */
static void destroy_mailbox(struct sip_mailbox *mailbox)
{
if (mailbox->mailbox)
@@ -2965,6 +2967,7 @@
ast_free(mailbox);
}
+/*! Destroy all peer-related mailbox subscriptions */
static void clear_peer_mailboxes(struct sip_peer *peer)
{
struct sip_mailbox *mailbox;
@@ -3220,7 +3223,10 @@
/*! \brief Locate peer by name or ip address
* This is used on incoming SIP message to find matching peer on ip
- or outgoing message to find matching peer on name */
+ or outgoing message to find matching peer on name
+ \note Avoid using this function in new functions if there's a way to avoid it, i
+ since it causes a database lookup or a traversal of the in-memory peer list.
+*/
static struct sip_peer *find_peer(const char *peer, struct sockaddr_in *sin, int realtime)
{
struct sip_peer *p = NULL;
@@ -3331,7 +3337,10 @@
}
/*! \brief Create address structure from peer reference.
- * return -1 on error, 0 on success.
+ * This function copies data from peer to the dialog, so we don't have to look up the peer
+ * again from memory or database during the life time of the dialog.
+ *
+ * \return -1 on error, 0 on success.
*/
static int create_addr_from_peer(struct sip_pvt *dialog, struct sip_peer *peer)
{
@@ -3371,9 +3380,9 @@
ast_udptl_destroy(dialog->udptl);
dialog->udptl = NULL;
}
- do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE );
-
- if (dialog->rtp) {
+ do_setnat(dialog, ast_test_flag(&dialog->flags[0], SIP_NAT) & SIP_NAT_ROUTE);
+
+ if (dialog->rtp) { /* Audio */
ast_rtp_setdtmf(dialog->rtp, ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833);
ast_rtp_setdtmfcompensate(dialog->rtp, ast_test_flag(&dialog->flags[1], SIP_PAGE2_RFC2833_COMPENSATE));
ast_rtp_set_rtptimeout(dialog->rtp, peer->rtptimeout);
@@ -3383,14 +3392,14 @@
ast_rtp_codec_setpref(dialog->rtp, &dialog->prefs);
dialog->autoframing = peer->autoframing;
}
- if (dialog->vrtp) {
+ if (dialog->vrtp) { /* Video */
ast_rtp_setdtmf(dialog->vrtp, 0);
ast_rtp_setdtmfcompensate(dialog->vrtp, 0);
ast_rtp_set_rtptimeout(dialog->vrtp, peer->rtptimeout);
ast_rtp_set_rtpholdtimeout(dialog->vrtp, peer->rtpholdtimeout);
ast_rtp_set_rtpkeepalive(dialog->vrtp, peer->rtpkeepalive);
}
- if (dialog->trtp) {
+ if (dialog->trtp) { /* Realtime text */
ast_rtp_setdtmf(dialog->trtp, 0);
ast_rtp_setdtmfcompensate(dialog->trtp, 0);
ast_rtp_set_rtptimeout(dialog->trtp, peer->rtptimeout);
@@ -3407,43 +3416,46 @@
ast_string_field_set(dialog, mohinterpret, peer->mohinterpret);
ast_string_field_set(dialog, tohost, peer->tohost);
ast_string_field_set(dialog, fullcontact, peer->fullcontact);
- if (!dialog->initreq.headers && !ast_strlen_zero(peer->fromdomain)) {
- char *tmpcall;
- char *c;
- tmpcall = ast_strdupa(dialog->callid);
- c = strchr(tmpcall, '@');
- if (c) {
- *c = '\0';
- ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
- }
- }
+ ast_string_field_set(dialog, context, peer->context);
dialog->outboundproxy = obproxy_get(dialog, peer);
+ dialog->callgroup = peer->callgroup;
+ dialog->pickupgroup = peer->pickupgroup;
+ dialog->allowtransfer = peer->allowtransfer;
+ dialog->jointnoncodeccapability = dialog->noncodeccapability;
+ dialog->rtptimeout = peer->rtptimeout;
+ dialog->maxcallbitrate = peer->maxcallbitrate;
if (ast_strlen_zero(dialog->tohost))
ast_string_field_set(dialog, tohost, ast_inet_ntoa(dialog->sa.sin_addr));
- if (!ast_strlen_zero(peer->fromdomain))
+ if (!ast_strlen_zero(peer->fromdomain)) {
ast_string_field_set(dialog, fromdomain, peer->fromdomain);
- if (!ast_strlen_zero(peer->fromuser))
+ if (!dialog->initreq.headers) {
+ char *c;
+ char *tmpcall = ast_strdupa(dialog->callid);
+
+ c = strchr(tmpcall, '@');
+ if (c) {
+ *c = '\0';
+ ast_string_field_build(dialog, callid, "%s@%s", tmpcall, peer->fromdomain);
+ }
+ }
+ }
+ if (!ast_strlen_zero(peer->fromuser))
ast_string_field_set(dialog, fromuser, peer->fromuser);
if (!ast_strlen_zero(peer->language))
ast_string_field_set(dialog, language, peer->language);
- dialog->callgroup = peer->callgroup;
- dialog->pickupgroup = peer->pickupgroup;
- dialog->allowtransfer = peer->allowtransfer;
+
/* Set timer T1 to RTT for this peer (if known by qualify=) */
/* Minimum is settable or default to 100 ms */
if (peer->maxms && peer->lastms)
dialog->timer_t1 = peer->lastms < global_t1min ? global_t1min : peer->lastms;
+
if ((ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_RFC2833) ||
(ast_test_flag(&dialog->flags[0], SIP_DTMF) == SIP_DTMF_AUTO))
dialog->noncodeccapability |= AST_RTP_DTMF;
else
dialog->noncodeccapability &= ~AST_RTP_DTMF;
- dialog->jointnoncodeccapability = dialog->noncodeccapability;
- ast_string_field_set(dialog, context, peer->context);
- dialog->rtptimeout = peer->rtptimeout;
if (peer->call_limit)
ast_set_flag(&dialog->flags[0], SIP_CALL_LIMIT);
- dialog->maxcallbitrate = peer->maxcallbitrate;
return 0;
}
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