From fa3c8f94e088036836d12a622b7f738d8dd48cc5 Mon Sep 17 00:00:00 2001
From: Kevin Harwell <kharwell@digium.com>
Date: Wed, 8 Apr 2020 14:33:47 -0500
Subject: [PATCH] chan_pjsip: digit_begin - constant DTMF tone if RTP is not
setup yet
If chan_pjsip is configured for DTMF_RFC_4733, and the core triggers a
digit begin before media, or rtp has been setup then it's possible the
outgoing channel will hear a constant DTMF tone upon answering.
This happens because when there is no media, or rtp chan_pjsip notifies
the core to initiate inband DTMF. However, upon digit end if media, and
rtp become available then chan_pjsip does not notify the core to stop
inband DTMF. Thus the tone continues playing.
This patch makes it so chan_pjsip only notifies the core to start
inband DTMF in only the required cases. Now if there is no media, or
rtp availabe upon digit begin chan_pjsip does nothing, but tells the
core it handled it.
ASTERISK-28817 #close
Change-Id: I0dbea9fff444a2595fb18c64b89653e90d2f6eb5
---
channels/chan_pjsip.c | 33 ++++++++++++++++++---------------
1 file changed, 18 insertions(+), 15 deletions(-)
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 590843e611..5df39c3bd6 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -2163,20 +2163,23 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct ast_sip_session_media *media;
- int res = 0;
media = channel->session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
switch (channel->session->dtmf) {
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
- return -1;
+ return 0;
}
ast_rtp_instance_dtmf_begin(media->rtp, digit);
break;
case AST_SIP_DTMF_AUTO:
- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+ if (!media || !media->rtp) {
+ return 0;
+ }
+
+ if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
return -1;
}
@@ -2191,13 +2194,12 @@ static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
- res = -1;
- break;
+ return -1;
default:
break;
}
- return res;
+ return 0;
}
struct info_dtmf_data {
@@ -2284,7 +2286,6 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct ast_sip_session_media *media;
- int res = 0;
if (!channel || !channel->session) {
/* This happens when the channel is hungup while a DTMF digit is playing. See ASTERISK-28086 */
@@ -2298,8 +2299,9 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
case AST_SIP_DTMF_AUTO_INFO:
{
if (!media || !media->rtp) {
- return -1;
+ return 0;
}
+
if (ast_rtp_instance_dtmf_mode_get(media->rtp) != AST_RTP_DTMF_MODE_NONE) {
ast_debug(3, "Told to send end of digit on Auto-Info channel %s RFC4733 negotiated so using it.\n", ast_channel_name(ast));
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
@@ -2337,28 +2339,29 @@ static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned in
}
case AST_SIP_DTMF_RFC_4733:
if (!media || !media->rtp) {
- return -1;
+ return 0;
}
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
break;
case AST_SIP_DTMF_AUTO:
- if (!media || !media->rtp || (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND)) {
+ if (!media || !media->rtp) {
+ return 0;
+ }
+
+ if (ast_rtp_instance_dtmf_mode_get(media->rtp) == AST_RTP_DTMF_MODE_INBAND) {
return -1;
}
ast_rtp_instance_dtmf_end_with_duration(media->rtp, digit, duration);
break;
-
-
case AST_SIP_DTMF_NONE:
break;
case AST_SIP_DTMF_INBAND:
- res = -1;
- break;
+ return -1;
}
- return res;
+ return 0;
}
static void update_initial_connected_line(struct ast_sip_session *session)
--
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