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codec_negotiation: Implement outgoing_call_offer_pref

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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
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7
main/stream.c
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@@ -186,6 +186,13 @@ struct ast_format_cap *ast_stream_get_formats(const struct ast_stream *stream)
return stream->formats;
}
int ast_stream_get_format_count(const struct ast_stream *stream)
{
ast_assert(stream != NULL);
return stream->formats ? ast_format_cap_count(stream->formats) : 0;
}
void ast_stream_set_formats(struct ast_stream *stream, struct ast_format_cap *caps)
{
ast_assert(stream != NULL);