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 a4cd452246
)
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@ -167,6 +167,14 @@ static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
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.incoming_request = chan_pjsip_incoming_ack,
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};
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static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
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static struct ast_sip_session_supplement chan_pjsip_prack_supplement = {
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.method = "PRACK",
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.priority = AST_SIP_SUPPLEMENT_PRIORITY_CHANNEL,
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.incoming_request = chan_pjsip_incoming_prack,
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};
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/*! \brief Function called by RTP engine to get local audio RTP peer */
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static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
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{
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@ -1564,13 +1572,22 @@ static int send_topology_change_refresh(void *data)
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{
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struct topology_change_refresh_data *refresh_data = data;
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struct ast_sip_session *session = refresh_data->session;
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enum ast_channel_state state = ast_channel_state(session->channel);
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enum ast_sip_session_refresh_method method = AST_SIP_SESSION_REFRESH_METHOD_INVITE;
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int ret;
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SCOPE_ENTER(3, "%s: %s\n", ast_sip_session_get_name(session),
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ast_str_tmp(256, ast_stream_topology_to_str(refresh_data->media_state->topology, &STR_TMP)));
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/* See RFC 6337, especially section 3.2: If the early media SDP was sent reliably, we are allowed
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* to send UPDATEs. Only relevant for AST_STATE_RINGING and AST_STATE_RING - if the channel is UP,
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* re-INVITES can be sent.
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*/
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if (session->early_confirmed && (state == AST_STATE_RINGING || state == AST_STATE_RING)) {
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method = AST_SIP_SESSION_REFRESH_METHOD_UPDATE;
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}
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ret = ast_sip_session_refresh(session, NULL, NULL, on_topology_change_response,
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AST_SIP_SESSION_REFRESH_METHOD_INVITE, 1, refresh_data->media_state);
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method, 1, refresh_data->media_state);
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refresh_data->media_state = NULL;
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topology_change_refresh_data_free(refresh_data);
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@ -3141,6 +3158,7 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
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pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
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if (sdp && sdp->body.ptr) {
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ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
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session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
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ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
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} else {
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ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
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@ -3161,6 +3179,7 @@ static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct
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ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
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ast_trace(1, "%s Method: %.*s Status: %d Queueing PROGRESS with SDP\n", ast_sip_session_get_name(session),
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(int)rdata->msg_info.cseq->method.name.slen, rdata->msg_info.cseq->method.name.ptr, status.code);
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session->early_confirmed = pjsip_100rel_is_reliable(rdata) == PJ_TRUE;
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ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
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}
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} else {
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@ -3195,6 +3214,18 @@ static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip
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SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
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}
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static int chan_pjsip_incoming_prack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
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{
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SCOPE_ENTER(3, "%s\n", ast_sip_session_get_name(session));
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if (pj_strcmp2(&rdata->msg_info.msg->line.req.method.name, "PRACK") == 0 &&
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pjmedia_sdp_neg_get_state(session->inv_session->neg) == PJMEDIA_SDP_NEG_STATE_DONE) {
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session->early_confirmed = 1;
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}
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SCOPE_EXIT_RTN_VALUE(0, "%s\n", ast_sip_session_get_name(session));
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}
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static int update_devstate(void *obj, void *arg, int flags)
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{
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ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE,
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@ -3307,6 +3338,7 @@ static int load_module(void)
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ast_sip_session_register_supplement(&call_pickup_supplement);
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ast_sip_session_register_supplement(&pbx_start_supplement);
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ast_sip_session_register_supplement(&chan_pjsip_ack_supplement);
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ast_sip_session_register_supplement(&chan_pjsip_prack_supplement);
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if (pjsip_channel_cli_register()) {
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ast_log(LOG_ERROR, "Unable to register PJSIP Channel CLI\n");
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@ -3326,6 +3358,7 @@ end:
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ao2_cleanup(pjsip_uids_onhold);
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pjsip_uids_onhold = NULL;
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ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
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ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
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ast_sip_session_unregister_supplement(&pbx_start_supplement);
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ast_sip_session_unregister_supplement(&chan_pjsip_supplement_response);
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ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
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@ -3355,6 +3388,7 @@ static int unload_module(void)
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ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
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ast_sip_session_unregister_supplement(&pbx_start_supplement);
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ast_sip_session_unregister_supplement(&chan_pjsip_ack_supplement);
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ast_sip_session_unregister_supplement(&chan_pjsip_prack_supplement);
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ast_sip_session_unregister_supplement(&call_pickup_supplement);
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ast_sip_unregister_service(&refer_callback_module);
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@ -231,6 +231,8 @@ struct ast_sip_session {
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unsigned int ended_while_deferred:1;
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/*! Whether to pass through hold and unhold using re-invites with recvonly and sendrecv */
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unsigned int moh_passthrough:1;
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/*! Whether early media state has been confirmed through PRACK */
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unsigned int early_confirmed:1;
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/*! DTMF mode to use with this session, from endpoint but can change */
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enum ast_sip_dtmf_mode dtmf;
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/*! Initial incoming INVITE Request-URI. NULL otherwise. */
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