chan_pjsip: Allow topology/session refreshes in early media state (#74)

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

diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 473e2b5dcc85c9c7fe408043d9a815edf8d658ec..8582af93ef36ebfb8baecda6f3276c714150de97 100644 (file)
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -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);

diff --git a/include/asterisk/res_pjsip_session.h b/include/asterisk/res_pjsip_session.h
index 471ecd462b80d7410c1070fe5e3e58bad6e2454e..a21ac4b066301ff509a110f92f781613910951bd 100644 (file)
--- a/include/asterisk/res_pjsip_session.h
+++ b/include/asterisk/res_pjsip_session.h
@@ -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. */