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codec negotiation: add incoming_call_offer_prefs option

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Add a new option, incoming_call_offer_pref, to res_pjsip endpoints that
specifies the preferred order of codecs after receiving an offer.

This patch does the following:

  Adds a new enumeration, ast_sip_call_codec_pref, used by the the new
configuration option that's added to the endpoint media structure.

  Adds a new ast_sip_session_caps structure that's set for each session media
object.

  Creates a new file, res_pjsip_session_caps that "implements" the new
structure and option, and is compiled into the res_pjsip_session library.

ASTERISK-28756 #close

Change-Id: I35e7a2a0c236cfb6bd9cdf89539f57a1ffefc76f
This commit is contained in:
Kevin Harwell
2020-02-24 12:47:46 -06:00
parent 87fda066ea
commit 06dada3f01
11 changed files with 484 additions and 20 deletions
Download Patch File Download Diff File Expand all files Collapse all files

99
res/res_pjsip_sdp_rtp.c
View File

@@ -56,6 +56,7 @@
#include "asterisk/res_pjsip.h"
#include "asterisk/res_pjsip_session.h"
#include "asterisk/res_pjsip_session_caps.h"
/*! \brief Scheduler for RTCP purposes */
static struct ast_sched_context *sched;
@@ -373,6 +374,81 @@ static void get_codecs(struct ast_sip_session *session, const struct pjmedia_sdp
}
}
static int apply_cap_to_bundled(struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
struct ast_stream *asterisk_stream, const struct ast_format_cap *joint)
{
if (!joint) {
return -1;
}
ast_stream_set_formats(asterisk_stream, (struct ast_format_cap *)joint);
/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
if (session_media_transport != session_media && session_media->bundled) {
int index;
for (index = 0; index < ast_format_cap_count(joint); ++index) {
struct ast_format *format = ast_format_cap_get_format(joint, index);
int rtp_code;
/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
* things as the format is guaranteed to have a payload already.
*/
rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
ao2_ref(format, -1);
}
}
return 0;
}
static const struct ast_format_cap *set_incoming_call_offer_cap(
struct ast_sip_session *session, struct ast_sip_session_media *session_media,
const struct pjmedia_sdp_media *stream)
{
const struct ast_format_cap *incoming_call_offer_cap;
struct ast_format_cap *remote;
struct ast_rtp_codecs codecs = AST_RTP_CODECS_NULL_INIT;
int fmts = 0;
remote = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!remote) {
ast_log(LOG_ERROR, "Failed to allocate %s incoming remote capabilities\n",
ast_codec_media_type2str(session_media->type));
return NULL;
}
/* Get the peer's capabilities*/
get_codecs(session, stream, &codecs, session_media);
ast_rtp_codecs_payload_formats(&codecs, remote, &fmts);
incoming_call_offer_cap = ast_sip_session_join_incoming_call_offer_cap(
session, session_media, remote);
ao2_ref(remote, -1);
if (!incoming_call_offer_cap) {
ast_rtp_codecs_payloads_destroy(&codecs);
return NULL;
}
/*
* Setup rx payload type mapping to prefer the mapping
* from the peer that the RFC says we SHOULD use.
*/
ast_rtp_codecs_payloads_xover(&codecs, &codecs, NULL);
ast_rtp_codecs_payloads_copy(&codecs,
ast_rtp_instance_get_codecs(session_media->rtp), session_media->rtp);
ast_rtp_codecs_payloads_destroy(&codecs);
return incoming_call_offer_cap;
}
static int set_caps(struct ast_sip_session *session,
struct ast_sip_session_media *session_media,
struct ast_sip_session_media *session_media_transport,
@@ -432,25 +508,7 @@ static int set_caps(struct ast_sip_session *session,
ast_rtp_codecs_payloads_copy(&codecs, ast_rtp_instance_get_codecs(session_media->rtp),
session_media->rtp);
ast_stream_set_formats(asterisk_stream, joint);
/* If this is a bundled stream then apply the payloads to RTP instance acting as transport to prevent conflicts */
if (session_media_transport != session_media && session_media->bundled) {
int index;
for (index = 0; index < ast_format_cap_count(joint); ++index) {
struct ast_format *format = ast_format_cap_get_format(joint, index);
int rtp_code;
/* Ensure this payload is in the bundle group transport codecs, this purposely doesn't check the return value for
* things as the format is guaranteed to have a payload already.
*/
rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(session_media->rtp), 1, format, 0);
ast_rtp_codecs_payload_set_rx(ast_rtp_instance_get_codecs(session_media_transport->rtp), rtp_code, format);
ao2_ref(format, -1);
}
}
apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream, joint);
if (session->channel && ast_sip_session_is_pending_stream_default(session, asterisk_stream)) {
ast_channel_lock(session->channel);
@@ -1420,7 +1478,8 @@ static int negotiate_incoming_sdp_stream(struct ast_sip_session *session,
session_media->remotely_held_changed = 1;
}
if (set_caps(session, session_media, session_media_transport, stream, 1, asterisk_stream)) {
if (apply_cap_to_bundled(session_media, session_media_transport, asterisk_stream,
set_incoming_call_offer_cap(session, session_media, stream))) {
return 0;
}