pjsip: Add rtp_timeout and rtp_timeout_hold endpoint options.
[asterisk/asterisk.git] / channels / chan_pjsip.c
index 09cec54..d3ba0a2 100644 (file)
@@ -625,6 +625,8 @@ static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
                return f;
        }
 
+       ast_rtp_instance_set_last_rx(media->rtp, time(NULL));
+
        if (f->frametype != AST_FRAME_VOICE) {
                return f;
        }