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rtpengine
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fix always-transcode not being honoured when only codec-mask was used

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reported in #664

Change-Id: I2cdfe405c61d48513a01393c5306fe4c875325ee
(cherry picked from commit 8f3c309c7f)
changes/19/28019/1
Richard Fuchs 7 years ago
parent adb65f53b8
commit a582399eb3
3 changed files with 149 additions and 3 deletions
Show Stats Download Patch File Download Diff File

31
daemon/codec.c
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@ -101,6 +101,7 @@ static struct codec_handler codec_handler_stub_ssrc = {
static void __handler_shutdown(struct codec_handler *handler) {
free_ssrc_hash(&handler->ssrc_hash);
handler->kernelize = 0;
handler->transcoder = 0;
}
static void __codec_handler_free(void *pp) {
@ -145,6 +146,8 @@ static void __make_transcoder(struct codec_handler *handler, struct rtp_payload_
assert(source->payload_type == handler->source_pt.payload_type);
// don't reset handler if it already matches what we want
if (!handler->transcoder)
goto reset;
if (rtp_payload_type_cmp(source, &handler->source_pt))
goto reset;
if (rtp_payload_type_cmp(dest, &handler->dest_pt))
@ -164,6 +167,7 @@ reset:
handler->source_pt = *source;
handler->dest_pt = *dest;
handler->func = handler_func_transcode;
handler->transcoder = 1;
handler->ssrc_hash = create_ssrc_hash_full(__ssrc_handler_transcode_new, handler);
@ -236,13 +240,34 @@ void codec_handlers_update(struct call_media *receiver, struct call_media *sink,
}
}
// similarly, if the sink receive a codec that the receiver can't send, it's also transcoding
if (MEDIA_ISSET(sink, TRANSCODE)) {
// similarly, if the sink can receive a codec that the receiver can't send, it's also transcoding
// XXX these blocks should go into their own functions
if (MEDIA_ISSET(sink, TRANSCODE) && !sink_transcoding) {
for (GList *l = sink->codecs_prefs_recv.head; l; l = l->next) {
struct rtp_payload_type *pt = l->data;
if (!g_hash_table_lookup(receiver->codec_names_recv, &pt->encoding))
GQueue *recv_pts = g_hash_table_lookup(receiver->codec_names_recv, &pt->encoding);
if (!recv_pts) {
sink_transcoding = 1;
goto done;
}
// even if the receiver can receive the same codec that the sink can
// send, we might still have it configured as a transcoder due to
// always-transcode in the offer
for (GList *k = recv_pts->head; k; k = k->next) {
// XXX codec_handlers can be converted to g_direct_hash table
int pt = GPOINTER_TO_INT(k->data);
struct codec_handler *ch_recv =
g_hash_table_lookup(sink->codec_handlers, &pt);
if (!ch_recv)
continue;
if (ch_recv->transcoder) {
sink_transcoding = 1;
goto done;
}
}
}
done:;
}
// stop transcoding if we've determined that we don't need it

1
include/codec.h
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@ -24,6 +24,7 @@ struct codec_handler {
struct rtp_payload_type dest_pt;
codec_handler_func *func;
int kernelize:1;
int transcoder:1;
struct ssrc_hash *ssrc_hash;
};

120
t/auto-daemon-tests.pl
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@ -1034,6 +1034,126 @@ SDP
# codec masking gh#664
new_call;
offer('gh 664 codec masking plain', { ICE => 'remove', replace => [qw(origin session-connection)],
flags => [qw(codec-mask-opus codec-mask-G722 codec-strip-G7221)] }, <<SDP);
v=0
o=- 1545997027 1 IN IP4 198.51.100.1
s=tester
t=0 0
m=audio 2000 RTP/AVP 120 8 0 101
c=IN IP4 198.51.100.1
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
----------------------------------
v=0
o=- 1545997027 1 IN IP4 203.0.113.1
s=tester
t=0 0
m=audio PORT RTP/AVP 8 0 101
c=IN IP4 203.0.113.1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:PORT
SDP
answer('gh 664 codec masking plain', { ICE => 'remove', replace => [qw(origin session-connection)] }, <<SDP);
v=0
o=- 1545997027 1 IN IP4 198.51.100.3
s=tester
c=IN IP4 198.51.100.3
t=0 0
m=audio 2002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--------------------------------------
v=0
o=- 1545997027 1 IN IP4 203.0.113.1
s=tester
c=IN IP4 203.0.113.1
t=0 0
m=audio PORT RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:PORT
SDP
new_call;
offer('gh 664 codec masking a/t', { ICE => 'remove', replace => [qw(origin session-connection)],
flags => [qw(codec-mask-opus codec-mask-G722 codec-strip-G7221 always-transcode)] }, <<SDP);
v=0
o=- 1545997027 1 IN IP4 198.51.100.1
s=tester
t=0 0
m=audio 2000 RTP/AVP 120 8 0 101
c=IN IP4 198.51.100.1
a=rtpmap:120 opus/48000/2
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
----------------------------------
v=0
o=- 1545997027 1 IN IP4 203.0.113.1
s=tester
t=0 0
m=audio PORT RTP/AVP 8 0 101
c=IN IP4 203.0.113.1
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:PORT
SDP
answer('gh 664 codec masking a/t', { ICE => 'remove', replace => [qw(origin session-connection)] }, <<SDP);
v=0
o=- 1545997027 1 IN IP4 198.51.100.3
s=tester
c=IN IP4 198.51.100.3
t=0 0
m=audio 2002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--------------------------------------
v=0
o=- 1545997027 1 IN IP4 203.0.113.1
s=tester
c=IN IP4 203.0.113.1
t=0 0
m=audio PORT RTP/AVP 120 8 0 101
a=rtpmap:120 opus/48000/2
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
a=fmtp:101 0-15
a=sendrecv
a=rtcp:PORT
SDP
END {
if ($rtpe_pid) {
kill('INT', $rtpe_pid) or die;

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