@ -143,23 +143,36 @@
# include "asterisk.h"
# include <math.h>
# include "asterisk/channel.h"
# include "asterisk/frame.h"
# include "asterisk/module.h"
# include "asterisk/rtp_engine.h"
# include <math.h> /* for sqrt, MAX */
# include <sched.h> /* for sched_yield */
# include <sys/time.h> /* for timeval */
# include <time.h> /* for time_t */
# include "asterisk/_private.h" /* for ast_rtp_engine_init prototype */
# include "asterisk/astobj2.h" /* for ao2_cleanup, ao2_ref, etc */
# include "asterisk/channel.h" /* for ast_channel_name, etc */
# include "asterisk/codec.h" /* for ast_codec_media_type2str, etc */
# include "asterisk/format.h" /* for ast_format_cmp, etc */
# include "asterisk/format_cache.h" /* for ast_format_adpcm, etc */
# include "asterisk/format_cap.h" /* for ast_format_cap_alloc, etc */
# include "asterisk/json.h" /* for ast_json_ref, etc */
# include "asterisk/linkedlists.h" /* for ast_rtp_engine::<anonymous>, etc */
# include "asterisk/lock.h" /* for ast_rwlock_unlock, etc */
# include "asterisk/logger.h" /* for ast_log, ast_debug, etc */
# include "asterisk/manager.h"
# include "asterisk/options.h"
# include "asterisk/astobj2.h"
# include "asterisk/pbx.h"
# include "asterisk/translate.h"
# include "asterisk/netsock2.h"
# include "asterisk/_private.h"
# include "asterisk/framehook.h"
# include "asterisk/stasis.h"
# include "asterisk/json.h"
# include "asterisk/stasis_channels.h"
# include "asterisk/module.h" /* for ast_module_unref, etc */
# include "asterisk/netsock2.h" /* for ast_sockaddr_copy, etc */
# include "asterisk/options.h" /* for ast_option_rtpptdynamic */
# include "asterisk/pbx.h" /* for pbx_builtin_setvar_helper */
# include "asterisk/res_srtp.h" /* for ast_srtp_res */
# include "asterisk/rtp_engine.h" /* for ast_rtp_codecs, etc */
# include "asterisk/stasis.h" /* for stasis_message_data, etc */
# include "asterisk/stasis_channels.h" /* for ast_channel_stage_snapshot, etc */
# include "asterisk/strings.h" /* for ast_str_append, etc */
# include "asterisk/time.h" /* for ast_tvdiff_ms, ast_tvnow */
# include "asterisk/translate.h" /* for ast_translate_available_formats */
# include "asterisk/utils.h" /* for ast_free, ast_strdup, etc */
# include "asterisk/vector.h" /* for AST_VECTOR_GET, etc */
struct ast_srtp_res * res_srtp = NULL ;
struct ast_srtp_policy_res * res_srtp_policy = NULL ;
@ -2301,6 +2314,48 @@ static void add_static_payload(int map, struct ast_format *format, int rtp_code)
}
}
/* http://www.iana.org/assignments/rtp-parameters
* RFC 3551 , Section 3 : " [...] applications which need to define more
* than 32 dynamic payload types MAY bind codes below 96 , in which case
* it is RECOMMENDED that unassigned payload type numbers be used
* first " . Updated by RFC 5761, Section 4: " [ . . . ] values in the range
* 64 - 95 MUST NOT be used [ to avoid conflicts with RTCP ] " . Summaries:
* https : //tools.ietf.org/html/draft-roach-mmusic-unified-plan#section-3.2.1.2
* https : //tools.ietf.org/html/draft-wu-avtcore-dynamic-pt-usage#section-3
*/
if ( map < 0 ) {
for ( x = MAX ( ast_option_rtpptdynamic , 35 ) ; x < = AST_RTP_PT_LAST_REASSIGN ; + + x ) {
if ( ! static_RTP_PT [ x ] ) {
map = x ;
break ;
}
}
}
/* Yet, reusing mappings below 35 is not supported in Asterisk because
* when Compact Headers are activated , no rtpmap is send for those below
* 35. If you want to use 35 and below
* A ) do not use Compact Headers ,
* B ) remove that code in chan_sip / res_pjsip , or
* C ) add a flag that this RTP Payload Type got reassigned dynamically
* and requires a rtpmap even with Compact Headers enabled .
*/
if ( map < 0 ) {
for ( x = MAX ( ast_option_rtpptdynamic , 20 ) ; x < 35 ; + + x ) {
if ( ! static_RTP_PT [ x ] ) {
map = x ;
break ;
}
}
}
if ( map < 0 ) {
for ( x = MAX ( ast_option_rtpptdynamic , 0 ) ; x < 20 ; + + x ) {
if ( ! static_RTP_PT [ x ] ) {
map = x ;
break ;
}
}
}
if ( map < 0 ) {
if ( format ) {
ast_log ( LOG_WARNING , " No Dynamic RTP mapping available for format %s \n " ,