[asterisk-commits] oej: branch 1.4 r47015 - /branches/1.4/channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Thu Nov  2 13:56:30 2006
New Revision: 47015

URL: http://svn.digium.com/view/asterisk?rev=47015&view=rev
Log:
Move check for codec translation to sip_call() instead of in add_sdp. No one bothers
with the result of add_sdp anyway... Yet...

Modified:
    branches/1.4/channels/chan_sip.c

Modified: branches/1.4/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/branches/1.4/channels/chan_sip.c?rev=47015&r1=47014&r2=47015&view=diff
==============================================================================
--- branches/1.4/channels/chan_sip.c (original)
+++ branches/1.4/channels/chan_sip.c Thu Nov  2 13:56:30 2006
@@ -1269,7 +1269,7 @@
 static void add_noncodec_to_sdp(const struct sip_pvt *p, int format, int sample_rate,
 				char **m_buf, size_t *m_size, char **a_buf, size_t *a_size,
 				int debug);
-static int add_sdp(struct sip_request *resp, struct sip_pvt *p);
+static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p);
 
 /*--- Authentication stuff */
 static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, int sipmethod, char *digest, int digest_len);
@@ -2783,16 +2783,21 @@
 	res = update_call_counter(p, INC_CALL_RINGING);
 	if ( res != -1 ) {
 		p->callingpres = ast->cid.cid_pres;
-		p->jointcapability = p->capability;
-		p->t38.jointcapability = p->t38.capability;
-		if (option_debug)
-			ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
-		transmit_invite(p, SIP_INVITE, 1, 2);
-		if (p->maxtime)
+		p->jointcapability = ast_translate_available_formats(p->capability, p->prefcodec);
+
+		/* If there are no audio formats left to offer, punt */
+		if (!(p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
+			ast_log(LOG_WARNING, "No audio format found to offer. Cancelling call to %s\n", p->username);
+			res = -1;
+		} else {
+			p->t38.jointcapability = p->t38.capability;
+			if (option_debug > 1)
+				ast_log(LOG_DEBUG,"Our T38 capability (%d), joint T38 capability (%d)\n", p->t38.capability, p->t38.jointcapability);
+			transmit_invite(p, SIP_INVITE, 1, 2);
+
 			/* Initialize auto-congest time */
-			p->initid = ast_sched_add(sched, p->maxtime * 4, auto_congest, p);
-		else 
-			p->initid = ast_sched_add(sched, SIP_TRANS_TIMEOUT, auto_congest, p);
+			p->initid = ast_sched_add(sched, p->maxtime ? (p->maxtime * 4) : SIP_TRANS_TIMEOUT, auto_congest, p);
+		}
 	}
 	return res;
 }
@@ -5966,7 +5971,7 @@
 }
 
 /*! \brief Add Session Description Protocol message */
-static int add_sdp(struct sip_request *resp, struct sip_pvt *p)
+static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
 {
 	int len = 0;
 	int alreadysent = 0;
@@ -6008,7 +6013,7 @@
 
 	if (!p->rtp) {
 		ast_log(LOG_WARNING, "No way to add SDP without an RTP structure\n");
-		return -1;
+		return AST_FAILURE;
 	}
 
 	/* Set RTP Session ID and version */
@@ -6032,14 +6037,8 @@
 		dest.sin_port = sin.sin_port;
 	}
 
-	/* Ok, let's start working with codec selection here */
-	capability = ast_translate_available_formats(p->jointcapability, p->prefcodec);
-
-	/* If there are no audio formats left to offer, punt */
-	if (!(capability & AST_FORMAT_AUDIO_MASK)) {
-		ast_log(LOG_WARNING, "No audio format found to offer.\n");
-		return -1;
-	}
+	capability = p->jointcapability;
+
 
 	if (option_debug > 1) {
 		char codecbuf[BUFSIZ];
@@ -6221,7 +6220,7 @@
 		ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
 	}
 
-	return 0;
+	return AST_SUCCESS;
 }
 
 /*! \brief Used for 200 OK and 183 early media */