[Asterisk-code-review] Unknown media codec resulting a call disconnect (asterisk[13])
Salah Ahmed
asteriskteam at digium.com
Tue Jan 14 22:01:46 CST 2020
Salah Ahmed has uploaded this change for review. ( https://gerrit.asterisk.org/c/asterisk/+/13595 )
Change subject: Unknown media codec resulting a call disconnect
......................................................................
Unknown media codec resulting a call disconnect
While an unknown rtp payloads are received, than call getting
dropped due to codec translation initiation failed.
ASTERISK-28691
Change-Id: I8d2e5c7b16227c2cb07b322c8704d43c189ffae8
---
M channels/chan_pjsip.c
1 file changed, 7 insertions(+), 2 deletions(-)
git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/95/13595/1
diff --git a/channels/chan_pjsip.c b/channels/chan_pjsip.c
index 33abe03..6474864 100644
--- a/channels/chan_pjsip.c
+++ b/channels/chan_pjsip.c
@@ -788,8 +788,13 @@
ao2_ref(caps, -1);
}
- ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format);
- ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
+ int wr_fmt_status = ast_set_write_format_path(ast, ast_channel_writeformat(ast), f->subclass.format);
+ int rd_fmt_status = ast_set_read_format_path(ast, ast_channel_readformat(ast), f->subclass.format);
+
+ if (wr_fmt_status == -1 || rd_fmt_status == -1) {
+ ast_frfree(f);
+ return &ast_null_frame;
+ }
if (ast_channel_is_bridged(ast)) {
ast_channel_set_unbridged_nolock(ast, 1);
--
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Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Change-Id: I8d2e5c7b16227c2cb07b322c8704d43c189ffae8
Gerrit-Change-Number: 13595
Gerrit-PatchSet: 1
Gerrit-Owner: Salah Ahmed <txrubel at gmail.com>
Gerrit-MessageType: newchange
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