Hello, This is on OpenSIPS 2.3, downloaded from git and compiled today.
An INVITE arrives over TLS with the following SDP: v=0 o=- 1492528621 1492528621 IN IP4 172.22.202.191 s=Polycom IP Phone c=IN IP4 172.22.202.191 t=0 0 m=audio 16852 RTP/SAVP 115 9 0 8 110 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=rtcp:16853 a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:[stripped] a=setup:actpass a=fingerprint:sha-1 [stripped] m=audio 16888 RTP/AVP 115 9 0 8 110 18 127 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:110 iLBC/8000 a=fmtp:110 mode=20 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:127 telephone-event/8000 a=rtcp:16889 I run codec_delete_expect_re(PCMU|PCMA|telephone-event) but it doesn't have any effect. The INVITE leaving after t_relay() over UDP to localhost on a different port is the same as when it came in (with the exception of the c= line because of rtpengine). At log_level=6 the only log entry I see is DBG:sipmsgops:create_codec_lumps: creating 0 streams I'm not sure where to go from here. - Jeff
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