SIP calls into Jigasi (from Freeswitch) -- fail on Jigasi-1.1-38, succeed on Jigasi-1.0-244

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?

I was having an odd issue with latest version of jigasi. add the following to sip-communicator.properties corrected it. might be worth a try.

net.java.sip.communicator.impl.protocol.sip.SKIP_REINVITE_ON_FOCUS_CHANGE_PROP=true

1 Like

@siacali does that ^ fixed it for you?

Thank you both for the speedy response! (pun intended). That seemed to have done the trick!

wish I could take credit for it, but I think @damencho had posted the answer in another thread. I came across it while looking to correct my issue! :smiley:

After dialing into a video conference on a soft phone on jigasi+freeswitch, there is no noise. After using the phone to dial in, the noise is very loud, but there is no noise after dialing from the phone to the softphone through freeswitch.

Can you help me see if there is a problem?