chan_sip: Filter pass-through audio/video formats away, again.

Instead of looking for pass-through formats in the list of transcodable
formats (which is going to find nothing), go through the result which
is going to be the jointcaps of the tech_pvt of the channel. Finally,
only with that list, ast_format_cap_remove(.) is going to succeed.

This restores the behaviour of Asterisk 1.8. However, it does not fix
ASTERISK_29282 because that issue report is about chan_sip and PJSIP.
Here, only chan_sip is fixed because PJSIP does not even call
ast_rtp_instance_available_formats -> ast_translate_available_format.

Change-Id: Icade2366ac2b82935b95a9981678c987da2e8c34

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 47db9e51d8abd246fc050945a407217425ad622a..52bcd5ca6d3c65b6970e966243d5e355381f220c 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13622,10 +13622,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
 
                /* Check if we need audio in this call */
                needaudio = ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_AUDIO);
-               if (!needaudio && p->outgoing_call) {
-                       /* p->caps are added conditionally, see below "Finally our remain..." */
-                       needaudio = ast_format_cap_has_type(p->caps, AST_MEDIA_TYPE_AUDIO);
-               }
 
                /* Check if we need video in this call */
                if ((ast_format_cap_has_type(tmpcap, AST_MEDIA_TYPE_VIDEO)) && !p->novideo) {
@@ -13756,7 +13752,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                /* Now, start adding audio codecs. These are added in this order:
                   - First what was requested by the calling channel
                   - Then our mutually shared capabilities, determined previous in tmpcap
-                  - Then preferences in order from sip.conf device config for this peer/user
                */
 
 
@@ -13800,27 +13795,6 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
                        ao2_ref(tmp_fmt, -1);
                }
 
-               /* Finally our remaining audio/video codecs */
-               for (x = 0; p->outgoing_call && x < ast_format_cap_count(p->caps); x++) {
-                       tmp_fmt = ast_format_cap_get_format(p->caps, x);
-
-                       if (ast_format_cap_iscompatible_format(alreadysent, tmp_fmt) != AST_FORMAT_CMP_NOT_EQUAL) {
-                               ao2_ref(tmp_fmt, -1);
-                               continue;
-                       }
-
-                       if (ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_AUDIO) {
-                               add_codec_to_sdp(p, tmp_fmt, &m_audio, &a_audio, debug, &min_audio_packet_size, &max_audio_packet_size);
-                       } else if (needvideo && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_VIDEO) {
-                               add_vcodec_to_sdp(p, tmp_fmt, &m_video, &a_video, debug, &min_video_packet_size);
-                       } else if (needtext && ast_format_get_type(tmp_fmt) == AST_MEDIA_TYPE_TEXT) {
-                               add_tcodec_to_sdp(p, tmp_fmt, &m_text, &a_text, debug, &min_text_packet_size);
-                       }
-
-                       ast_format_cap_append(alreadysent, tmp_fmt, 0);
-                       ao2_ref(tmp_fmt, -1);
-               }
-
                /* Now add DTMF RFC2833 telephony-event as a codec */
                for (x = 1LL; x <= AST_RTP_MAX; x <<= 1) {
                        if (!(p->jointnoncodeccapability & x))

diff --git a/main/translate.c b/main/translate.c
index 6648931b72237a9793d76c3bb50e6f65fc5c3615..a9665ae348cee990630cd348d660955bef6af7f0 100644 (file)
--- a/main/translate.c
+++ b/main/translate.c
@@ -1509,16 +1509,19 @@ static void check_translation_path(
        struct ast_format_cap *result, struct ast_format *src_fmt,
        enum ast_media_type type)
 {
-       int index, src_index = format2index(src_fmt);
+       int i;
+
+       if (ast_format_get_type(src_fmt) != type) {
+               return;
+       }
+
        /* For a given source format, traverse the list of
           known formats to determine whether there exists
           a translation path from the source format to the
           destination format. */
-       for (index = 0; (src_index >= 0) && index < cur_max_index; index++) {
-               struct ast_codec *codec = index2codec(index);
-               RAII_VAR(struct ast_format *, fmt, ast_format_create(codec), ao2_cleanup);
-
-               ao2_ref(codec, -1);
+       for (i = ast_format_cap_count(result) - 1; 0 <= i; i--) {
+               int index, src_index;
+               struct ast_format *fmt = ast_format_cap_get_format(result, i);
 
                if (ast_format_get_type(fmt) != type) {
                        continue;
@@ -1535,6 +1538,15 @@ static void check_translation_path(
                        continue;
                }
 
+               /* if this is a pass-through format, not in the source,
+                  we cannot transcode. Therefore, remove it from the result */
+               src_index = format2index(src_fmt);
+               index = format2index(fmt);
+               if (src_index < 0 || index < 0) {
+                       ast_format_cap_remove(result, fmt);
+                       continue;
+               }
+
                /* if we don't have a translation path from the src
                   to this format, remove it from the result */
                if (!matrix_get(src_index, index)->step) {