Merge of the "sdpcleanup" branch. Thanks to John Martin for a lot of tests
and some patches (all disclaimed). - Don't change RTP properties if we reject a re-INVITE - Don't add video to an outbound channel if there's no video on the inbound channel - Don't include video in the "preferred codec" list for codec selection - Clean up and document code that parses and adds SDP attachments Since we do not transcode video, we can't handle video the same way as audio. This is a bug fix patch. In future releases, we need to work on a solution for video negotiation, not codecs but formats and framerates instead. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32597 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej
parent
77abfdfd00
commit
4506e03f3d
11
channel.c
11
channel.c
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@ -539,7 +539,7 @@ char *ast_transfercapability2str(int transfercapability)
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}
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}
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/*! \brief Pick the best codec */
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/*! \brief Pick the best audio codec */
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int ast_best_codec(int fmts)
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{
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/* This just our opinion, expressed in code. We are asked to choose
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@ -572,7 +572,9 @@ int ast_best_codec(int fmts)
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/*! Down to G.723.1 which is proprietary but at least designed for voice */
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AST_FORMAT_G723_1,
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};
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/* Strip out video */
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fmts &= AST_FORMAT_AUDIO_MASK;
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/* Find the first preferred codec in the format given */
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for (x=0; x < (sizeof(prefs) / sizeof(prefs[0]) ); x++)
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@ -2614,6 +2616,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
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int fmt;
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int res;
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int foo;
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int videoformat = format & AST_FORMAT_VIDEO_MASK;
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if (!cause)
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cause = &foo;
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@ -2629,7 +2632,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
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continue;
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capabilities = chan->tech->capabilities;
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fmt = format;
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fmt = format & AST_FORMAT_AUDIO_MASK;
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res = ast_translator_best_choice(&fmt, &capabilities);
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if (res < 0) {
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ast_log(LOG_WARNING, "No translator path exists for channel type %s (native %d) to %d\n", type, chan->tech->capabilities, format);
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@ -2640,7 +2643,7 @@ struct ast_channel *ast_request(const char *type, int format, void *data, int *c
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if (!chan->tech->requester)
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return NULL;
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if (!(c = chan->tech->requester(type, capabilities, data, cause)))
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if (!(c = chan->tech->requester(type, capabilities | videoformat, data, cause)))
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return NULL;
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if (c->_state == AST_STATE_DOWN) {
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File diff suppressed because it is too large
Load Diff
10
frame.c
10
frame.c
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@ -1008,9 +1008,12 @@ int ast_codec_choose(struct ast_codec_pref *pref, int formats, int find_best)
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break;
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}
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}
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if(ret)
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if(ret & AST_FORMAT_AUDIO_MASK)
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return ret;
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if (option_debug > 3)
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ast_log(LOG_DEBUG, "Could not find preferred codec - %s\n", find_best ? "Going for the best codec" : "Returning zero codec");
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return find_best ? ast_best_codec(formats) : 0;
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}
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@ -1034,7 +1037,10 @@ void ast_parse_allow_disallow(struct ast_codec_pref *pref, int *mask, const char
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*mask &= ~format;
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}
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if (pref) {
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/* Set up a preference list for audio. Do not include video in preferences
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since we can not transcode video and have to use whatever is offered
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*/
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if (pref && (format & AST_FORMAT_AUDIO_MASK)) {
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if (strcasecmp(this, "all")) {
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if (allowing)
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ast_codec_pref_append(pref, format);
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@ -40,13 +40,13 @@ extern "C" {
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/* Codes for RTP-specific data - not defined by our AST_FORMAT codes */
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/*! DTMF (RFC2833) */
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#define AST_RTP_DTMF (1 << 0)
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#define AST_RTP_DTMF (1 << 0)
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/*! 'Comfort Noise' (RFC3389) */
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#define AST_RTP_CN (1 << 1)
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#define AST_RTP_CN (1 << 1)
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/*! DTMF (Cisco Proprietary) */
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#define AST_RTP_CISCO_DTMF (1 << 2)
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#define AST_RTP_CISCO_DTMF (1 << 2)
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/*! Maximum RTP-specific code */
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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#define AST_RTP_MAX AST_RTP_CISCO_DTMF
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#define MAX_RTP_PT 256
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@ -62,6 +62,9 @@ struct ast_rtp_protocol {
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AST_LIST_ENTRY(ast_rtp_protocol) list;
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};
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#define FLAG_3389_WARNING (1 << 0)
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typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *data);
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@ -71,7 +74,6 @@ typedef int (*ast_rtp_callback)(struct ast_rtp *rtp, struct ast_frame *f, void *
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* RTP session is defined on page 9 of RFC 3550: "An association among a set of participants communicating with RTP. A participant may be involved in multiple RTP sessions at the same time [...]"
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*
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*/
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/*! \brief The value of each payload format mapping: */
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struct rtpPayloadType {
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int isAstFormat; /*!< whether the following code is an AST_FORMAT */
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@ -23,7 +23,8 @@
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#ifndef _ASTERISK_TRANSLATE_H
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#define _ASTERISK_TRANSLATE_H
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#define MAX_FORMAT 32
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//#define MAX_FORMAT 15 /* Do not include video here */
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#define MAX_FORMAT 32 /* Do include video here */
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#if defined(__cplusplus) || defined(c_plusplus)
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extern "C" {
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7
rtp.c
7
rtp.c
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@ -1350,7 +1350,7 @@ int ast_rtp_make_compatible(struct ast_channel *dest, struct ast_channel *src, i
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return 1;
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}
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/*! \brief Make a note of a RTP paymoad type that was seen in a SDP "m=" line.
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/*! \brief Make a note of a RTP payload type that was seen in a SDP "m=" line.
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* By default, use the well-known value for this type (although it may
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* still be set to a different value by a subsequent "a=rtpmap:" line)
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*/
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@ -1359,9 +1359,8 @@ void ast_rtp_set_m_type(struct ast_rtp* rtp, int pt)
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if (pt < 0 || pt > MAX_RTP_PT)
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return; /* bogus payload type */
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if (static_RTP_PT[pt].code != 0) {
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if (static_RTP_PT[pt].code != 0)
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rtp->current_RTP_PT[pt] = static_RTP_PT[pt];
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}
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}
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/*! \brief Make a note of a RTP payload type (with MIME type) that was seen in
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@ -2245,7 +2244,7 @@ int ast_rtp_write(struct ast_rtp *rtp, struct ast_frame *_f)
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/* Make sure we have enough space for RTP header */
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if ((_f->frametype != AST_FRAME_VOICE) && (_f->frametype != AST_FRAME_VIDEO)) {
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ast_log(LOG_WARNING, "RTP can only send voice\n");
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ast_log(LOG_WARNING, "RTP can only send voice and video\n");
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return -1;
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}
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