TT#88603 don't remove transcoded codecs we added ourselves

closes #1058

Change-Id: I469b8e71d5a699c948a30cb2171467c267eec741

daemon/codec.c

@@ -968,8 +968,10 @@ void codec_handlers_update(struct call_media *receiver, struct call_media *sink,
 		if (MEDIA_ISSET(sink, TRANSCODE)) {
 			// if the other side is transcoding, we may come across a receiver entry
 			// (recv->recv) that wasn't originally offered (recv->send). we must eliminate
-			// those
-			if (!g_hash_table_lookup(receiver->codecs_send, &pt->payload_type)) {
+			// those, unless we added them ourselves for transcoding.
+			struct rtp_payload_type *recv_pt =
+				g_hash_table_lookup(receiver->codecs_send, &pt->payload_type);
+			if (!recv_pt && !pt->for_transcoding) {
 				ilog(LOG_DEBUG, "Eliminating transcoded codec " STR_FORMAT,
 						STR_FMT(&pt->encoding_with_params));
 
@@ -2382,6 +2384,7 @@ void codec_rtp_payload_types(struct call_media *media, struct call_media *other_
 		pt = codec_add_payload_type(codec, media);
 		if (!pt)
 			continue;
+		pt->for_transcoding = 1;
 
 		if (__codec_synth_transcode_options(pt, flags, media))
 			continue;

lib/rtplib.h

@@ -29,6 +29,8 @@ struct rtp_payload_type {
 	int bitrate;
 
 	const codec_def_t *codec_def;
+
+	int for_transcoding:1;
 };
 
 

t/auto-daemon-tests.pl

@@ -36,6 +36,113 @@ my ($sock_a, $sock_b, $sock_c, $sock_d, $port_a, $port_b, $ssrc, $resp,
 
 
 
+# GH 1058
+
+new_call;
+
+offer('missing codec in re-invite', {
+		ICE => 'remove',
+		flags => ["codec-mask-all", "codec-strip-telephone-event", "codec-transcode-PCMU", "codec-transcode-G722", "codec-transcode-t38", "codec-offer-telephone-event", "port-latching"],
+		'to-tag' => tt(),
+	}, <<SDP);
+v=0
+o=dev 623840 205550 IN IP4 8.8.8.61
+s=SIP Media Capabilities
+c=IN IP4 8.8.8.61
+t=0 0
+m=audio 6304 RTP/AVP 0 8 3 101
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:3 GSM/8000
+a=rtpmap:101 telephone-event/8000
+a=fmtp:101 0-15
+a=sendrecv
+a=maxptime:20
+--------------------------------------
+v=0
+o=dev 623840 205550 IN IP4 8.8.8.61
+s=SIP Media Capabilities
+c=IN IP4 203.0.113.1
+t=0 0
+m=audio PORT RTP/AVP 101 0 9
+a=maxptime:20
+a=rtpmap:101 telephone-event/8000
+a=rtpmap:0 PCMU/8000
+a=rtpmap:9 G722/8000
+a=fmtp:101 0-15
+a=sendrecv
+a=rtcp:PORT
+SDP
+
+answer('missing codec in re-invite', {
+		ICE => 'remove',
+		flags => ["trust-address", "symmetric-codecs"],
+	}, <<SDP);
+v=0
+o=dev 5418 9648 IN IP4 8.8.8.60
+s=SIP Call
+c=IN IP4 8.8.8.60
+t=0 0
+m=audio 6004 RTP/AVP 9 101
+c=IN IP4 8.8.8.60
+a=rtpmap:9 G722/8000
+a=fmtp:9 bitrate=64
+a=rtpmap:101 telephone-event/8000
+a=fmtp:101 0-15
+--------------------------------------
+v=0
+o=dev 5418 9648 IN IP4 8.8.8.60
+s=SIP Call
+c=IN IP4 8.8.8.60
+t=0 0
+m=audio PORT RTP/AVP 101 0 8 3
+c=IN IP4 203.0.113.1
+a=rtpmap:101 telephone-event/8000
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:3 GSM/8000
+a=fmtp:101 0-15
+a=sendrecv
+a=rtcp:PORT
+SDP
+
+offer('missing codec in re-invite', {
+		ICE => 'remove',
+		flags => ["codec-mask-all", "codec-strip-telephone-event", "codec-transcode-PCMU", "codec-transcode-G722", "codec-transcode-t38", "codec-offer-telephone-event", "port-latching"],
+		'to-tag' => tt(),
+	}, <<SDP);
+v=0
+o=dev 623840 205550 IN IP4 8.8.8.61
+s=SIP Media Capabilities
+c=IN IP4 8.8.8.61
+t=0 0
+m=audio 6304 RTP/AVP 0 8 3 101
+a=rtpmap:0 PCMU/8000
+a=rtpmap:8 PCMA/8000
+a=rtpmap:3 GSM/8000
+a=rtpmap:101 telephone-event/8000
+a=fmtp:101 0-15
+a=sendrecv
+a=maxptime:20
+--------------------------------------
+v=0
+o=dev 623840 205550 IN IP4 8.8.8.61
+s=SIP Media Capabilities
+c=IN IP4 203.0.113.1
+t=0 0
+m=audio PORT RTP/AVP 101 0 9
+a=maxptime:20
+a=rtpmap:101 telephone-event/8000
+a=rtpmap:0 PCMU/8000
+a=rtpmap:9 G722/8000
+a=fmtp:101 0-15
+a=sendrecv
+a=rtcp:PORT
+SDP
+
+
+
+
 # DTLS-reverse flag
 
 new_call;