[Sofia-sip-devel] UAS INVITE processing

2006-12-11 Thread Roman Filonenko
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

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[Sofia-sip-devel] Problem with HOLD/RESUME

2006-12-11 Thread [EMAIL PROTECTED]
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

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[Sofia-sip-devel] sresolv module questions

2006-12-11 Thread Colin Whittaker
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..


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[Sofia-sip-devel] anyone ever see windows clients lie about their bitrate?

2006-12-11 Thread Mike Frantzen

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
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[Sofia-sip-devel] nta or nua

2006-12-11 Thread Yuling Duff

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 
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[Sofia-sip-devel] hello!

2006-12-11 Thread Caesar Caligula

hi:
  I am a student learning SIP. I have a question, can sofia-sip support 
the windows os?


regards:)

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