Merged revisions 293493 via svnmerge from

https://origsvn.digium.com/svn/asterisk/branches/1.8 [^]

........
  r293493 | twilson | 2010-11-01 09:58:00 -0500 (Mon, 01 Nov 2010) | 14 lines

  Only offer codecs both sides support for directmedia

  When using directmedia, Asterisk needs to limit the codecs offered to just
  the ones that both sides recognize, otherwise they may end up sending audio
  that the other side doesn't understand.

  (closes issue 0017403)
  Reported by: one47
  Patches:
        sip_codecs_simplified4 uploaded by one47 (license 23)
  Tested by: one47, falves11

  Review: https://reviewboard.asterisk.org/r/967/ [^]
........

Backporting a bugfix that should have been included.

git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.2@302049 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 238421eccad75acbfacec7dce99e852fbaea0d94..c1be361e842454909e3253f2cd73845018e4b1ce 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10193,6 +10193,7 @@ static void get_our_media_address(struct sip_pvt *p, int needvideo,
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
 {
        int alreadysent = 0;
+       int doing_directmedia = FALSE;
 
        struct sockaddr_in sin;
        struct sockaddr_in vsin;
@@ -10254,6 +10255,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
        }
 
        if (add_audio) {
+               doing_directmedia = (p->redirip.sin_addr.s_addr && p->redircodecs) ? TRUE : FALSE;
                /* Check if we need video in this call */
                if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
                        if (p->vrtp) {
@@ -10284,6 +10286,16 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
        snprintf(connection, sizeof(connection), "c=IN IP4 %s\r\n", ast_inet_ntoa(dest.sin_addr));
 
        if (add_audio) {
+               if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR) {
+                       hold = "a=recvonly\r\n";
+                       doing_directmedia = FALSE;
+               } else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE) {
+                       hold = "a=inactive\r\n";
+                       doing_directmedia = FALSE;
+               } else {
+                       hold = "a=sendrecv\r\n";
+               }
+
                capability = p->jointcapability;
 
                /* XXX note, Video and Text are negated - 'true' means 'no' */
@@ -10291,6 +10303,11 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                          p->novideo ? "True" : "False", p->notext ? "True" : "False");
                ast_debug(1, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
 
+               if (doing_directmedia) {
+                       capability &= p->redircodecs;
+                       ast_debug(1, "** Our native-bridge filtered capablity: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability));
+               }
+
                /* Check if we need audio */
                if (capability & AST_FORMAT_AUDIO_MASK)
                        needaudio = TRUE;
@@ -10336,13 +10353,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 
                ast_str_append(&m_audio, 0, "m=audio %d RTP/AVP", ntohs(dest.sin_port));
 
-               if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_ONEDIR)
-                       hold = "a=recvonly\r\n";
-               else if (ast_test_flag(&p->flags[1], SIP_PAGE2_CALL_ONHOLD) == SIP_PAGE2_CALL_ONHOLD_INACTIVE)
-                       hold = "a=inactive\r\n";
-               else
-                       hold = "a=sendrecv\r\n";
-
                /* Now, start adding audio codecs. These are added in this order:
                   - First what was requested by the calling channel
                   - Then preferences in order from sip.conf device config for this peer/user