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• local RTP port unreachable when using mjsip/jmf

  - by brian_d

  Hello, I create a sip session with mjsip to an external voip provider. Then I transmit a test wav file over rtp to the provider using RtpManager. The program runs with no errors and I answer the sip call. However, no audio is transmitted. When I diagnose the network traffic with wireshark, I see a bunch of RTP traffic from my localhost (behind some kind of nat) to the voip provider and nothing back. After a while I get the ICMP error "Destination unreachable (Port unreachable)" from the provider to my localhost. The software linphone works using the same localhost and voip provider - though it is using a different sip stack. Any suggestions? Thanks

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• Videoconference using Flash and SIP

  - by Júlio Santos

  The front-end will be Flash, to run in a browser and have access to the camera. I must use SIP to control the sessions. How could I do this? Will a Red5 server and a MjSip sever do the trick? As in i'd use MjSip to setup the session and warn users about calls, and Red5 to stream the video and audio? Any suggestions? Note: only 1-on-1 conference is required.

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• The best way to impliment a SIP VOIP client for blackberry

  - by Afzal

  How to implement a SIP/VOIP client for Blackberry smart phones? where to begin? should I start with mjsip? is there any sample code available specific for Blackberry? Thanks in advance, Afzal

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