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Wierd RTP Issue

I have a peculiar RTP issue. I’m experimenting with Jitsi as a softphone on one of my desktop Windows machines. That machine can either be connected to Asterisk via an VPN connection (with a static IP address) or not (via NAT). When it’s connected via NAT, all is OK.

When it’s connected with VPN, the following occurs:

The voice path inbound to Jitsi works fine when Jitsi originates the call, no matter what it’s calling.

The voice path inbound to Jitsi works fine when it’s called from another SIP
device.

The voice path inbound to Jitsi is silent when it’s called from something on the other side of a PRI via DAHDI.

I’ve run Wireshark on my desktop and see the RTP packets coming at the same rate and protocol (g711) in all the cases and “sip set debug peer xyz”
doesn’t shed any light on the situation (the SDP data looks similar in the working and non-worknig cases).

Does anybody have any ideas what to look at next?

Jabber/XMPP

Hi all,

Friday at 12 Noon EDT, we’ll be talking to Emil Ivov of Jitsi.org
(formerly SIP Communicator) and Thiago Rocha Camargo (of Nimbuzz)
about Jabber, something the Asterisk community is becoming more
interested in by the day. Join us to learn more about Jabber and SIP
or to share your knowledge and experience. As always, the VUC
discussion includes people from very diverse backgrounds, so it should
be a unique approach to the subject.

All the info to connect is on this page: http://vuc.me

- SIP:200901@login.zipdx.com (g722, g711)
- Skype:vuc.me and ld.vuc.me
- IRC #vuc
- PSTN +15672522286
- iNum +883510012394882
- gtalk voipusersconference@gmail.com

During the conference hours, there’s a widget to join on the above
page as well as an mp3 stream link.

Join us!

:r