[Sofia-sip-devel] UAS INVITE processing
RFC3261, section 13.3.1 ("Processing of INVITE") has a following bullet: If the request is an INVITE that contains an Expires header field, the UAS core sets a timer for the number of seconds indicated in the header field value. When the timer fires, the invitation is considered to be expired. If the invitation expires before the UAS has generated a final response, a 487 (Request Terminated) response SHOULD be generated. I suppose that to implement the above, one needs to define a separate dialog usage (like nua_session_usage). Is that correct? Is the bullet planned for implementation? Best regards, Roman - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] Problem with HOLD/RESUME
Hello, I have a problem making Hold and Resume with my sip calls. I'm using sofia-sip darcs version. When I want to make a hold, I call nua_invite with NUTAG_HOLD(1) to send a re-invite with "sendonly" added in the media attibute of the SDP parameters, at this stage all work well. But when I Resume the call with NUTAG_HOLD(0), the re-invite is sent but without media attribute added, normally it must add "sendrecv" to the media attributes no? Did I make something wrong to hold and resume calls or are there a problem about this in sofia-sip??? Thank you to help me. emilie - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] sresolv module questions
I've started using DNS, and it appears that I must specify the FQDN for the proxy or registrar, and that the DNS server must have any entry. Is there a way to get the sresolv module to use the /etc/hosts file ? I've setup nsswitch.conf to use /etc/hosts first: hosts: files dns But it doesn't appear to use /etc/hosts at all. Also, it appears that it is not using the default domain in the /etc/resolve.conf Is this true ? Colin.. - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV ___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] anyone ever see windows clients lie about their bitrate?
I realize this isn't strictly Sofia-SIP but it affects the RTP side of things built on Sofia. I've tried two windows clients, SJPhone and X-Lite and both lie about their bitrate. Both are advertising a PCMU/8000 codec. I watched the RTP packets in a sniffer and ran the numbers: SJPhone is actually sending 8400 bits per second (52.5 packets per second) and X-Lite is sending 7800 bits per second. Both are running on the same windows box with a cheap embedded dell sound board. The RTP stream with 8400 bits per second is causing a nasty 5% audio latency when discontinuous transmission is turned off; the latency grows at a rate of 3 seconds per minute since the windows client is sending 20ms worth of audio every 19.05ms. The extra buffering is eventually fatal to the audio stream on my N770 when the DSP runs out of buffers (I think). I've also seen similar affects from the only other windows box I've ever tried but didn't have tcpdump handy to run the numbers myself. Has anyone ever seen anything like this? I've been told that getting short accurate audio samples with the Windows APIs but can't confirm that. I'm looking at dynamically calculating a RTP streams real bitrate and resampling it if it's too far off the claimed bitrate. That seems rather kludgy. thanks, .mike - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] nta or nua
Hi, I started look into sofia-sip but not sure between nua and nta. If our application only need a sip stack to process a few sip messages like invite, cancel, ack, options and bye, is nta module is sufficient? If so how to use it. Has anyone used just nta for their applications? Thanks for your help. YD - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel
[Sofia-sip-devel] hello!
hi: I am a student learning SIP. I have a question, can sofia-sip support the windows os? regards:) _ 免费下载 MSN Explorer: http://explorer.msn.com/lccn/ - Take Surveys. Earn Cash. Influence the Future of IT Join SourceForge.net's Techsay panel and you'll get the chance to share your opinions on IT & business topics through brief surveys - and earn cash http://www.techsay.com/default.php?page=join.php&p=sourceforge&CID=DEVDEV___ Sofia-sip-devel mailing list Sofia-sip-devel@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/sofia-sip-devel