Merged revisions 376131 via svnmerge from

file:///srv/subversion/repos/asterisk/trunk

................
  r376131 | file | 2012-11-11 11:15:47 -0600 (Sun, 11 Nov 2012) | 16 lines
  
  Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default.
  
  With ICE support enabled in chan_sip and a large number of interfaces on the system it was
  possible for the produced SDP to be truncated due to some fixed size buffers. These buffers
  have now been changed so they will dynamically grow as needed.
  
  ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience
  for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is
  no longer enabled by default there.
  
  (closes issue ASTERISK-20643)
  Reported by: coopvr
  ........
  
  Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376136 65c4cc65-6c06-0410-ace0-fbb531ad65f3

1 files changed, 6 insertions, 9 deletions

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 9059669f0..c939df955 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12927,9 +12927,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 	struct ast_str *m_video = ast_str_alloca(256);  /* Media declaration line for video */
 	struct ast_str *m_text = ast_str_alloca(256);   /* Media declaration line for text */
 	struct ast_str *m_modem = ast_str_alloca(256);  /* Media declaration line for modem */
-	struct ast_str *a_audio = ast_str_alloca(1024); /* Attributes for audio */
-	struct ast_str *a_video = ast_str_alloca(1024); /* Attributes for video */
-	struct ast_str *a_text = ast_str_alloca(1024);  /* Attributes for text */
+	struct ast_str *a_audio = ast_str_create(256); /* Attributes for audio */
+	struct ast_str *a_video = ast_str_create(256); /* Attributes for video */
+	struct ast_str *a_text = ast_str_create(256);  /* Attributes for text */
 	struct ast_str *a_modem = ast_str_alloca(1024); /* Attributes for modem */
 	const char *a_crypto = NULL;
 	const char *v_a_crypto = NULL;
@@ -13193,11 +13193,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 
 			add_dtls_to_sdp(p->rtp, &a_audio);
 		}
-
-		if (m_audio->len - m_audio->used < 2 || m_video->len - m_video->used < 2 ||
-		    m_text->len - m_text->used < 2 || a_text->len - a_text->used < 2 ||
-		    a_audio->len - a_audio->used < 2 || a_video->len - a_video->used < 2)
-			ast_log(LOG_WARNING, "SIP SDP may be truncated due to undersized buffer!!\n");
 	}
 
 	if (add_t38) {
@@ -13376,6 +13371,9 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 	ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmpcap));
 
 add_sdp_cleanup:
+	ast_free(a_text);
+	ast_free(a_video);
+	ast_free(a_audio);
 	alreadysent = ast_format_cap_destroy(alreadysent);
 	tmpcap = ast_format_cap_destroy(tmpcap);
 
@@ -31227,7 +31225,6 @@ static int reload_config(enum channelreloadreason reason)
 	ast_set_flag(&global_flags[0], SIP_DTMF_RFC2833);    /*!< Default DTMF setting: RFC2833 */
 	ast_set_flag(&global_flags[0], SIP_DIRECT_MEDIA);    /*!< Allow re-invites */
 	ast_set_flag(&global_flags[2], SIP_PAGE3_NAT_AUTO_RPORT); /*!< Default to nat=auto_force_rport */
-	ast_set_flag(&global_flags[2], SIP_PAGE3_ICE_SUPPORT); /*!< Default to enabling ICE support */
 	ast_copy_string(default_engine, DEFAULT_ENGINE, sizeof(default_engine));
 	ast_copy_string(default_parkinglot, DEFAULT_PARKINGLOT, sizeof(default_parkinglot));