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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
Download Patch File Download Diff File Expand all files Collapse all files

54
include/asterisk/res_pjsip.h
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@@ -511,22 +511,40 @@ enum ast_sip_session_redirect {
/*!
* \brief Incoming/Outgoing call offer/answer joint codec preference.
*
* The default is INTERSECT ALL LOCAL.
*/
enum ast_sip_call_codec_pref {
/*! Two bits for merge */
/*! Intersection of local and remote */
AST_SIP_CALL_CODEC_PREF_INTERSECT = 1 << 0,
/*! Union of local and remote */
AST_SIP_CALL_CODEC_PREF_UNION = 1 << 1,
/*! Two bits for filter */
/*! No filter */
AST_SIP_CALL_CODEC_PREF_ALL = 1 << 2,
/*! Only the first */
AST_SIP_CALL_CODEC_PREF_FIRST = 1 << 3,
/*! Two bits for preference and sort */
/*! Prefer, and order by local values */
AST_SIP_CALL_CODEC_PREF_LOCAL,
/*! Prefer, and order by local values (intersection) */
AST_SIP_CALL_CODEC_PREF_LOCAL_LIMIT,
/*! Prefer, and order by local values (top/first only) */
AST_SIP_CALL_CODEC_PREF_LOCAL_SINGLE,
AST_SIP_CALL_CODEC_PREF_LOCAL = 1 << 4,
/*! Prefer, and order by remote values */
AST_SIP_CALL_CODEC_PREF_REMOTE,
/*! Prefer, and order by remote values (intersection) */
AST_SIP_CALL_CODEC_PREF_REMOTE_LIMIT,
/*! Prefer, and order by remote values (top/first only) */
AST_SIP_CALL_CODEC_PREF_REMOTE_SINGLE,
AST_SIP_CALL_CODEC_PREF_REMOTE = 1 << 5,
};
/*!
* \brief Returns true if the preference is set in the parameter
* \since 18.0.0
*
* \param param A ast_flags struct with one or more of enum ast_sip_call_codec_pref set
* \param codec_pref The last component of one of the enum values
* \retval 1 if the enum value is set
* \retval 0 if not
*/
#define ast_sip_call_codec_pref_test(__param, __codec_pref) (!!(ast_test_flag( &__param, AST_SIP_CALL_CODEC_PREF_ ## __codec_pref )))
/*!
* \brief Session timers options
*/
@@ -769,7 +787,9 @@ struct ast_sip_endpoint_media_configuration {
/*! Enable webrtc settings and defaults */
unsigned int webrtc;
/*! Codec preference for an incoming offer */
enum ast_sip_call_codec_pref incoming_call_offer_pref;
struct ast_flags incoming_call_offer_pref;
/*! Codec preference for an outgoing offer */
struct ast_flags outgoing_call_offer_pref;
};
/*!
@@ -3222,6 +3242,18 @@ int ast_sip_dtmf_to_str(const enum ast_sip_dtmf_mode dtmf,
*/
int ast_sip_str_to_dtmf(const char *dtmf_mode);
/*!
* \brief Convert the call codec preference flags to a string
* \since 18.0.0
*
* \param pref the call codec preference setting
*
* \returns a constant string with either the setting value or 'unknown'
* \note Don't try to free the string!
*
*/
const char *ast_sip_call_codec_pref_to_str(struct ast_flags pref);
/*!
* \brief Transport shutdown monitor callback.
* \since 13.18.0