chan_pjsip: Allow topology/session refreshes in early media state

With this change, session modifications in the early media state are
possible if the SDP was sent reliably and confirmed by a PRACK. For
details, see RFC 6337, escpecially section 3.2.

Resolves: #73
(cherry picked from commit 1a7dada804)
This commit is contained in:
Maximilian Fridrich 2023-05-10 16:38:04 +02:00 committed by Asterisk Development Team
parent c5c89ed377
commit 173c433274
2 changed files with 37 additions and 1 deletions

View File

@ -167,6 +167,14 @@ static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
.incoming_request = chan_pjsip_incoming_ack,
};
static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
static struct ast_sip_session_supplement chan_pjsip_prack_supplement = {
.method = "PRACK",
.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
.incoming_request = chan_pjsip_incoming_prack,
};
/*! \brief Function called by RTP engine to get local audio RTP peer */
static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
@ -1564,13 +1572,22 @@ static int send_topology_change_refresh(void *data)
{
struct topology_change_refresh_data *refresh_data = data;
struct ast_sip_session *session = refresh_data->session;
enum ast_channel_state state = ast_channel_state(session->channel);
enum ast_sip_session_refresh_method method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
int ret;
SCOPE_ENTER(3, "%s: %s\n", ast_sip_session_get_name(session),
ast_str_tmp(256, ast_stream_topology_to_str(refresh_data->media_state->topology, &STR_TMP)));
/* See RFC 6337, especially section 3.2: If the early media SDP was sent reliably, we are allowed
* to send UPDATEs. Only relevant for AST_STATE_RINGING and AST_STATE_RING - if the channel is UP,
* re-INVITES can be sent.
*/
if (session->early_confirmed && (state == AST_STATE_RINGING || state == AST_STATE_RING)) {
method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
}
ret = ast_sip_session_refresh(session, NULL, NULL, on_topology_change_response,
AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, refresh_data->media_state);
method, 1, refresh_data->media_state);
refresh_data->media_state = NULL;
topology_change_refresh_data_free(refresh_data);
@ -3141,6 +3158,7 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
if (sdp && sdp->body.ptr) {
ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
} else {
ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
@ -3161,6 +3179,7 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
}
} else {
@ -3195,6 +3214,18 @@ static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
}
static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
session->early_confirmed = 1;
}
SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
}
static int update_devstate(void *obj, void *arg, int flags)
{
ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE,
@ -3307,6 +3338,7 @@ static int load_module(void)
ast_sip_session_register_supplement(&call_pickup_supplement);
ast_sip_session_register_supplement(&pbx_start_supplement);
ast_sip_session_register_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_register_supplement(&chan_pjsip_prack_supplement);
if (pjsip_channel_cli_register()) {
ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
@ -3326,6 +3358,7 @@ end:
ao2_cleanup(pjsip_uids_onhold);
pjsip_uids_onhold = NULL;
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
@ -3355,6 +3388,7 @@ static int unload_module(void)
ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
ast_sip_session_unregister_supplement(&call_pickup_supplement);
ast_sip_unregister_service(&refer_callback_module);

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@ -231,6 +231,8 @@ struct ast_sip_session {
unsigned int ended_while_deferred:1;
/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
unsigned int moh_passthrough:1;
/*! Whether early media state has been confirmed through PRACK */
unsigned int early_confirmed:1;
/*! DTMF mode to use with this session, from endpoint but can change */
enum ast_sip_dtmf_mode dtmf;
/*! Initial incoming INVITE Request-URI. NULL otherwise. */