[asterisk-bugs] [JIRA] (ASTERISK-29842) Do not change 180 Ringing to 183 Progress even if early_media already enabled

[Vit Bohacek (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29842?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=260751#comment-260751 ] 

Vit Bohacek edited comment on ASTERISK-29842 at 11/30/22 8:25 AM:
------------------------------------------------------------------

Hello,
this patch doesn't solve the case when Ringing comes with SDP in it's body.

I use asterisk 18.14.0
it is necessary to add ast_sip_get_allow_sending_180_after_183() condition into chan_pjsip_incoming_response - chan_pjsip.c   

example: 
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
        struct pjsip_status_line status = rdata->msg_info.msg->line.status;
        SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);

        if (!session->channel) {
                SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
        }

        ast_log(LOG_WARNING,"chan_pjsip_incoming_response %i\n",status.code);
        switch (status.code) {
        case 180: {
                pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
                if (sdp && sdp->body.ptr) {
+                        if(ast_sip_get_allow_sending_180_after_183()) {
+                               ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
+                                ast_queue_control(session->channel, AST_CONTROL_RINGING);
+                        } else {
+                                ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
+                                ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
+                        }
−                         ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
−                         ast_queue_control(session->channel, AST_CONTROL_PROGRESS);

                } else {
                        ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
                        ast_queue_control(session->channel, AST_CONTROL_RINGING);
                }




was (Author: cofein):
Hello,
this patch doesn't solve the case when Ringing comes with SDP in it's body.

I use asterisk 18.14.0
it is necessary to add ast_sip_get_allow_sending_180_after_183() condition into chan_pjsip_incoming_response - chan_pjsip.c   

example: 
static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
        struct pjsip_status_line status = rdata->msg_info.msg->line.status;
        SCOPE_ENTER(3, "%s: Status: %d\n", ast_sip_session_get_name(session), status.code);

        if (!session->channel) {
                SCOPE_EXIT_RTN("%s: No channel\n", ast_sip_session_get_name(session));
        }

        ast_log(LOG_WARNING,"chan_pjsip_incoming_response %i\n",status.code);
        switch (status.code) {
        case 180: {
                pjsip_rdata_sdp_info *sdp = pjsip_rdata_get_sdp_info(rdata);
                if (sdp && sdp->body.ptr) {
+                        if(ast_sip_get_allow_sending_180_after_183()) {
+                               ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
+                                ast_queue_control(session->channel, AST_CONTROL_RINGING);
+                        } else {
+                                ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
+                                ast_queue_control(session->channel, AST_CONTROL_PROGRESS);
+                        }
-                         ast_trace(-1, "%s: Queueing PROGRESS\n", ast_sip_session_get_name(session));
-                         ast_queue_control(session->channel, AST_CONTROL_PROGRESS);

                } else {
                        ast_trace(-1, "%s: Queueing RINGING\n", ast_sip_session_get_name(session));
                        ast_queue_control(session->channel, AST_CONTROL_RINGING);
                }



> Do not change 180 Ringing to 183 Progress even if early_media already enabled
> -----------------------------------------------------------------------------
>
>                 Key: ASTERISK-29842
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29842
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 16.23.0, 18.9.0, 19.1.0
>            Reporter: Mark Petersen
>            Assignee: Mark Petersen
>      Target Release: 16.26.0, 18.12.0, 19.4.0, 20.0.0
>
>
> Currently if we receive a 180 Ringing and we previously have received a 183 Progress with SDP, asterisk is send as 183 Progress with SDP
> but it should be send as 180 Ringing with SDP
> RFC-3960 3.2.  Ringing Tone Generation
> SIP User Agent (UA) could implement the following
>    local policy:
>       1. Unless a 180 (Ringing) response is received, never generate
>          local ringing.
>       2. If a 180 (Ringing) has been received but there are no incoming
>          media packets, generate local ringing.
>       3. If a 180 (Ringing) has been received and there are incoming
>          media packets, play them and do not generate local ringing.
>          Note that a 180 (Ringing) response means that the callee is
>          being alerted, and a UAS should send such a response if the
>          callee is being alerted, regardless of the status of the early
>          media session.



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