kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-09-02 19:16:15 +00:00
Code Issues Packages Projects Releases Wiki Activity

res_pjsip_sdp_rtp.c: Initial RTP inactivity check must consider the rtp_timeout setting.

Browse Source 
First rtp activity check was performed after 500ms regardless of the rtp_timeout setting. Having a call in ringing state for more than rtp_timeout and the first rtp package is received more than 500ms after sdp negotiation and before the rtp_timeout, erronously caused the call to be hungup. Changed to perform the first rtp inactivity check after the timeout setting preventing calls to be disconnected before the rtp_timeout has elapsed since sdp negotiation.

Fixes #710

(cherry picked from commit d74266fe05)
This commit is contained in:
Henrik Liljedahl
2024-04-11 12:56:55 +02:00
committed by Asterisk Development Team
parent 229f3035f0
commit f580b8b5f1
1 changed files with 5 additions and 2 deletions
Download Patch File Download Diff File Expand all files Collapse all files

7
res/res_pjsip_sdp_rtp.c
View File

@@ -2101,6 +2101,7 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
enum ast_media_type media_type = session_media->type;
char host[NI_MAXHOST];
int res;
int rtp_timeout;
struct ast_sip_session_media *session_media_transport;
SCOPE_ENTER(1, "%s Stream: %s\n", ast_sip_session_get_name(session),
ast_str_tmp(128, ast_stream_to_str(asterisk_stream, &STR_TMP)));
@@ -2246,8 +2247,10 @@ static int apply_negotiated_sdp_stream(struct ast_sip_session *session,
ast_rtp_instance_set_timeout(session_media->rtp, session->endpoint->media.rtp.timeout_hold);
}
if (ast_rtp_instance_get_timeout(session_media->rtp)) {
session_media->timeout_sched_id = ast_sched_add_variable(sched, 500, rtp_check_timeout,
rtp_timeout = ast_rtp_instance_get_timeout(session_media->rtp);
if (rtp_timeout) {
session_media->timeout_sched_id = ast_sched_add_variable(sched, rtp_timeout*1000, rtp_check_timeout,
session_media, 1);
}