kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-03 11:25:35 +00:00
Code Issues Packages Projects Releases Wiki Activity

codec_negotiation: Implement outgoing_call_offer_pref

Browse Source 
Based on this new endpoint setting, a joint list of preferred codecs
between those received from the Asterisk core (remote), and those
specified in the endpoint's "allow" parameter (local) is created and
is used to create the outgoing SDP offer.

* Add outgoing_call_offer_pref to pjsip_configuration (endpoint)

* Add "call_direction" to res_pjsip_session.

* Update pjsip_session_caps.c to make the functions more generic
  so they could be used for both incoming and outgoing.

* Update ast_sip_session_create_outgoing to create the
  pending_media_state->topology with the results of
  ast_sip_session_create_joint_call_stream().

* The endpoint "preferred_codec_only" option now automatically sets
  AST_SIP_CALL_CODEC_PREF_FIRST in incoming_call_offer_pref.

* A helper function ast_stream_get_format_count() was added to
  streams to return the current count of formats.

ASTERISK-28777

Change-Id: Id4ec0b4a906c2ae5885bf947f101c59059935437
This commit is contained in:
George Joseph
2020-03-13 13:40:46 -06:00
committed by Friendly Automation
parent 57a457c26c
commit 2ee455958e
13 changed files with 408 additions and 297 deletions
Download Patch File Download Diff File Expand all files Collapse all files

46
res/res_pjsip_session.c
View File

@@ -468,8 +468,6 @@ static void session_media_dtor(void *obj)
ast_free(session_media->mid);
ast_free(session_media->remote_mslabel);
ao2_cleanup(session_media->caps);
}
struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_session *session,
@@ -528,12 +526,6 @@ struct ast_sip_session_media *ast_sip_session_media_state_add(struct ast_sip_ses
} else {
session_media->bundle_group = -1;
}
session_media->caps = ast_sip_session_caps_alloc();
if (!session_media->caps) {
ao2_ref(session_media, -1);
return NULL;
}
}
if (AST_VECTOR_REPLACE(&media_state->sessions, position, session_media)) {
@@ -2701,6 +2693,8 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
return NULL;
}
session->aor = ao2_bump(found_aor);
session->call_direction = AST_SIP_SESSION_OUTGOING_CALL;
ast_party_id_copy(&session->id, &endpoint->id.self);
if (ast_stream_topology_get_count(req_topology) > 0) {
@@ -2709,8 +2703,6 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
for (i = 0; i < ast_stream_topology_get_count(req_topology); ++i) {
struct ast_stream *req_stream;
struct ast_format_cap *req_cap;
struct ast_format_cap *joint_cap;
struct ast_stream *clone_stream;
req_stream = ast_stream_topology_get_stream(req_topology, i);
@@ -2719,39 +2711,12 @@ struct ast_sip_session *ast_sip_session_create_outgoing(struct ast_sip_endpoint
continue;
}
req_cap = ast_stream_get_formats(req_stream);
joint_cap = ast_format_cap_alloc(AST_FORMAT_CAP_FLAG_DEFAULT);
if (!joint_cap) {
clone_stream = ast_sip_session_create_joint_call_stream(session, req_stream);
if (!clone_stream || ast_stream_get_format_count(clone_stream) == 0) {
ast_stream_free(clone_stream);
continue;
}
ast_format_cap_get_compatible(req_cap, endpoint->media.codecs, joint_cap);
if (ast_stream_get_type(req_stream) == AST_MEDIA_TYPE_AUDIO) {
/*
* By appending codecs from the endpoint after compatible ones this
* guarantees that priority is given to those while also allowing
* translation to occur for non-compatible.
*/
ast_format_cap_append_from_cap(joint_cap,
endpoint->media.codecs, AST_MEDIA_TYPE_AUDIO);
}
if (!ast_format_cap_count(joint_cap)) {
ao2_ref(joint_cap, -1);
continue;
}
clone_stream = ast_stream_clone(req_stream, NULL);
if (!clone_stream) {
ao2_ref(joint_cap, -1);
continue;
}
ast_stream_set_formats(clone_stream, joint_cap);
ao2_ref(joint_cap, -1);
if (!session->pending_media_state->topology) {
session->pending_media_state->topology = ast_stream_topology_alloc();
if (!session->pending_media_state->topology) {
@@ -3351,6 +3316,7 @@ static void handle_new_invite_request(pjsip_rx_data *rdata)
#endif
return;
}
session->call_direction = AST_SIP_SESSION_INCOMING_CALL;
/*
* The current thread is supposed be the session serializer to prevent