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pjsip show channelstats: Prevent possible segfault when faxing

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Under rare circumstances, it's possible for the original audio
session in the active_media_state default_session to be corrupted
instead of removed when switching to the t38/image media session
during fax negotiation.  This can cause a segfault when a "pjsip
show channelstats" attempts to print that audio media session's
rtp statistics.  In these cases, the active_media_state
topology is correctly showing only a single t38/image stream
so we now check that there's an audio stream in the topology
before attempting to use the audio media session to get the rtp
statistics.

Resolves: #592
This commit is contained in:
George Joseph
2024-02-09 09:15:13 -07:00
committed by asterisk-org-access-app[bot]
parent 6871d1cdfc
commit 29a273618d
2 changed files with 21 additions and 3 deletions
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20
channels/pjsip/cli_commands.c
View File

@@ -343,6 +343,7 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
struct ast_sip_session *session;
struct ast_sip_session_media *media;
struct ast_rtp_instance_stats stats;
struct ast_stream *stream;
char *print_name = NULL;
char *print_time = alloca(32);
char codec_in_use[7];
@@ -359,16 +360,29 @@ static int cli_channelstats_print_body(void *obj, void *arg, int flags)
cpvt = ast_channel_tech_pvt(channel);
session = cpvt ? cpvt->session : NULL;
if (!session) {
if (!session
|| !session->active_media_state
|| !session->active_media_state->topology) {
ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->base->name);
ast_channel_unlock(channel);
ao2_cleanup(channel);
return 0;
}
stream = ast_stream_topology_get_first_stream_by_type(
session->active_media_state->topology, AST_MEDIA_TYPE_AUDIO);
if (!stream) {
ast_str_append(&context->output_buffer, 0, " %s no audio streams\n", snapshot->base->name);
ast_channel_unlock(channel);
ao2_cleanup(channel);
return 0;
}
media = session->active_media_state->default_session[AST_MEDIA_TYPE_AUDIO];
if (!media || !media->rtp) {
ast_str_append(&context->output_buffer, 0, " %s not valid\n", snapshot->base->name);
if (!media || media->type != AST_MEDIA_TYPE_AUDIO || !media->rtp) {
ast_str_append(&context->output_buffer, 0, " %s corrupted default audio session\n", snapshot->base->name);
ast_channel_unlock(channel);
ao2_cleanup(channel);
return 0;