SIP calls from Freeswitch to Jigasi-1.1-38 fail, but succeed on Jigasi-1.0-244. With nothing changed but the version of Jigasi, 1.0 gives me an SDP in the invite that looks like this:
m=audio 10008 RTP/AVP 0 8 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 v=0 o=9900-jitsi.org 0 0 IN IP4 10.1.1.120 s=- c=IN IP4 10.1.1.120 t=0 0 a=rtpmap:101 telephone-event/8000 m=video 0 RTP/AVP 104
This works fine.
While 1.1 sends an invite with an SDP that looks like this:
v=0 o=9900-jitsi.org 0 0 IN IP4 10.1.1.120 s=- c=IN IP4 10.1.1.120 t=0 0 m=audio 10009 RTP/AVP 9 0 8 96 3 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 opus/48000/2 a=fmtp:96 usedtx=1 a=ptime:20 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=extmap:1 urn:ietf:params:rtp-hdrext:csrc-audio-level a=extmap:2 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=rtcp-xr:voip-metrics m=video 10011 RTP/AVP 97 a=recvonly
With the invite issued by Jigasi 1.1, Freeswitch responds with a 491 Request Pending, which causes Jigasi to send an ACK, and immediately with no further reply from Freeswitch, a 488 Not Acceptable here. The call is taken down and fails to connect properly to the conference.
Does anyone know why? I’ve seen this discussed on another thread, which led me to try the older version. Are there any plans to fix this? Short of either hacking Jigasi so it sends less, or using an older version, is there any known workaround?