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 #17403)
  Reported by: one47
  Patches:
        sip_codecs_simplified4 uploaded by one47 (license 23)
  Tested by: one47, falves11

  Review: https://reviewboard.asterisk.org/r/967/
........

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302048 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 84dd415..749f7a2 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -10597,6 +10597,7 @@ static void get_crypto_attrib(struct sip_srtp *srtp, const char **a_crypto)
 static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int oldsdp, int add_audio, int add_t38)
 {
        format_t alreadysent = 0;
+       int doing_directmedia = FALSE;
 
        struct ast_sockaddr addr = { {0,} };
        struct ast_sockaddr vaddr = { {0,} };
@@ -10661,6 +10662,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
        }
 
        if (add_audio) {
+               doing_directmedia = (!ast_sockaddr_isnull(&p->redirip) && p->redircodecs) ? TRUE : FALSE;
                /* Check if we need video in this call */
                if ((p->jointcapability & AST_FORMAT_VIDEO_MASK) && !p->novideo) {
                        if (p->vrtp) {
@@ -10700,6 +10702,16 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                 ast_sockaddr_stringify_addr(&dest));
 
        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' */
@@ -10707,6 +10719,11 @@ 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;
@@ -10752,13 +10769,6 @@ 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/%s", ast_sockaddr_port(&dest),
                        a_crypto ? "SAVP" : "AVP");
 
-               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