[asterisk-bugs] [JIRA] (ASTERISK-27448) Add ability to send progress inband by setting a channel variable instead of fixed config per endpoint

[nappsoft (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27448?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

nappsoft updated ASTERISK-27448:
--------------------------------

    Description: 
I've faced a problem with several voip switches when using 180 Ringing messages after announcements sent over 183 Session Progress with SDP. The solution would be to use inband_progress=yes but this way progress will be sent as rtp for each call, so as well in cases in which it is not necessary.

So my idea was to set a channel variable PROGRESSINBAND to 1 in situations in which we use Progress() followed by Playback().

I've added some code to chan_pjsip.c which seems to work. I've opened a ticket here for two reasons: one: the "hack" could be usefull for someone else and second: I wanted to ask whether it is safe to do it the way I solved this or whether I'd need to lock channel->session->channel or something else first?

--- asterisk-13.18.0/channels/chan_pjsip.c	2017-10-30 16:33:07.000000000 +0100
+++ asterisk-13.18.0_mod/channels/chan_pjsip.c	2017-11-29 12:02:55.000000000 +0100
@@ -1335,6 +1335,15 @@
 				res = -1;
 			} else {
 				response_code = 180;
+
+				const char *ref_by_val;
+				ref_by_val = pbx_builtin_getvar_helper(channel->session->channel, "PROGRESSINBAND");
+				if (!ast_strlen_zero(ref_by_val)) {
+					if (ref_by_val[0]=='1') {
+						response_code = 183;
+						res = -1;
+					}
+				}
 			}
 		} else {
 			res = -1;


  was:
I faced a problem with several voip switches when using 180 Ringing messages after announcements sent over 183 Session Progress with SDP. The solution would be to use inband_progress=yes but this way progress will be sent as rtp for each call, so as well in cases in which it is not necessary.

So my idea was to set a channel variable PROGRESSINBAND to 1 in situations in which we use Progress() followed by Playback().

I've added some code to chan_pjsip.c which seems to work. I've opened a ticket here for two reasons: one: the "hack" could be usefull for someone else and second: I wanted to ask whether it is safe to do it the way I solved this or whether I'd need to lock channel->session->channel or something else first?

--- asterisk-13.18.0/channels/chan_pjsip.c	2017-10-30 16:33:07.000000000 +0100
+++ asterisk-13.18.0_mod/channels/chan_pjsip.c	2017-11-29 12:02:55.000000000 +0100
@@ -1335,6 +1335,15 @@
 				res = -1;
 			} else {
 				response_code = 180;
+
+				const char *ref_by_val;
+				ref_by_val = pbx_builtin_getvar_helper(channel->session->channel, "PROGRESSINBAND");
+				if (!ast_strlen_zero(ref_by_val)) {
+					if (ref_by_val[0]=='1') {
+						response_code = 183;
+						res = -1;
+					}
+				}
 			}
 		} else {
 			res = -1;



> Add ability to send progress inband by setting a channel variable instead of fixed config per endpoint
> ------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-27448
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27448
>             Project: Asterisk
>          Issue Type: Improvement
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.18.2
>            Reporter: nappsoft
>            Severity: Minor
>         Attachments: progressinband.diff
>
>
> I've faced a problem with several voip switches when using 180 Ringing messages after announcements sent over 183 Session Progress with SDP. The solution would be to use inband_progress=yes but this way progress will be sent as rtp for each call, so as well in cases in which it is not necessary.
> So my idea was to set a channel variable PROGRESSINBAND to 1 in situations in which we use Progress() followed by Playback().
> I've added some code to chan_pjsip.c which seems to work. I've opened a ticket here for two reasons: one: the "hack" could be usefull for someone else and second: I wanted to ask whether it is safe to do it the way I solved this or whether I'd need to lock channel->session->channel or something else first?
> --- asterisk-13.18.0/channels/chan_pjsip.c	2017-10-30 16:33:07.000000000 +0100
> +++ asterisk-13.18.0_mod/channels/chan_pjsip.c	2017-11-29 12:02:55.000000000 +0100
> @@ -1335,6 +1335,15 @@
>  				res = -1;
>  			} else {
>  				response_code = 180;
> +
> +				const char *ref_by_val;
> +				ref_by_val = pbx_builtin_getvar_helper(channel->session->channel, "PROGRESSINBAND");
> +				if (!ast_strlen_zero(ref_by_val)) {
> +					if (ref_by_val[0]=='1') {
> +						response_code = 183;
> +						res = -1;
> +					}
> +				}
>  			}
>  		} else {
>  			res = -1;



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