Re: [GnomeMeeting-devel-list] problem with incoming call (SIP)

[Damien Sandras <dsandras seconix com>]

Le mercredi 27 avril 2005 à 12:16 +0200, Damien Ciabrini a écrit :
> Damien Sandras wrote:
> > Le mercredi 27 avril 2005 à 11:11 +0200, Damien Ciabrini a écrit :
> > 
> > 
> >>Damien, thank you so much for the support ! It works :))
> >>
> >>Just for clarification: Now, connection and incoming INVITEs go to port 
> >>5060. It's fine. But what is the listen_port for ? I've just listened 
> >>with ethereal, and port listen_port (5080 for me now) is never used.
> >>
> > 
> > 
> > OPAL has a weird way to deal with things. I have asked to Robert and
> > Craig to change that.
> > 
> > Usually, all SIPPhones are doing this :
> > - they listen on port XYZ (5060)
> > - all their REGISTER/SUBSCRIBE/... requests are done with source port
> > XYZ, and the contact field contains the XYZ port
> > 
> > OPAL is working differently :
> > - it listens on port XYZ (5060)
> > - all REGISTER/SUBSCRIBE/INVITE/... requests are done with source port
> > XYZ+1 or +2 or +3 or ...., and the contact field contains the XYZ port.
> > 
> > For REGISTER requests, it will keep listening on the source port used
> > for the registration as long as the registration is active.
> > 
> > So, in your case, 5080 will never be used.
> > In a normal case, 5060 would always be used, and 5061 (XYZ+1, +2,
> > +3, ...) will only be used to refresh the REGISTER and receive answers
> > to it. That approach is giving problem with firewalls and NAT, because
> > the signalling is NOT symmetric.
> > 
> OK, so for what I understood, we force Opal to listen to port 5080 so 
> that registering can occurs on port 5060 instead of 5061. But then, what 
> kind of packet is Opal supposed to listen to on port 5080 ??
> 

Logically, incoming connections. But 5080 is not specified in the
contact field (bug), so you will receive them on 5060. The solution is
to fix opal so that it supports symmetric signalling.

> > I have asked them to change that as it is a core change in the behavior.
> > 
> > So I would say, keep using that workaround until another solution is
> > found. 
> That's fine for me.
> > 
> > Btw, I see you are using Wengo, why GnomeMeeting/X-Lite/kphone instead
> > of their opensource client? And why GnomeMeeting and not
> > kphone/linphone?
> Sure, here they are:
> 
> Cons of Wengo:
> 1. last time I checked their snapshot there was not sound support on 
> linux, only stubs to complete the compilation. I'm not sure if they're 
> goind to support OSS or ALSA.
> 2. I'm using Wengo because currently only them offer appealing prices 
> for calling PSTN in France. But I don't want to be locked in: for 
> instance they don't offer free voicemail as other SIP providers do (such 
> as sipgate). So it's definitely better to use a generic SIP client such 
> as Xlite instead of the Wengo client.
> 3. Support for codecs is limited.
> Pros of Wengo:
> 1. Maybe easy to configure on win32, but that's all for now :)
> 
> 
> Cons of X-lite:
> 1. Not open source. (really annoying when something doesn't work)
> 2. uses OSS instead of ALSA. This causes troubles with my bluetooth headset.
> Pros of X-lite:
> 1. Configuration is easy. Somebody before you has already spent time to 
> configure X-lite for your SIP provider :)
> 2. You see the codec used during a call.
> 
> Cons of kphone/linphone:
> 1. I'm a gnome user. ok that's a dumb reason :P
> 2. Friends trying kphone told me that there was a annoying lag during 
> the call. To be verified...
> 3. kphone asks for my Wengo password each tume I start it (annoying 
> because Wengo passwords are autogenerated)
> 4. Only one SIP identity in KPhone. too bad if you have many call plans 
> from several SIP providers.
> Pros:
> <none for me right now>
> 
> Cons of GM:
> 1. GUI is bad, especially if you just want to do VoIP.

What would you change? Have you tried the new UI from CVS?

> 2. I have not found a way to see what codec is used during a call.

tools->general history
or latest CVS.

> 3. Opal connection behavior :)
> 4. I have not found a way to say to GM "use this particular SIP proxy 
> for calling this particular contact" :(
> Pros:
> 1. Realtime stats on the quality of the call. You definitely trust 
> better the GM developpers for that.
> 2. Many codecs.
> 3. Not only for VoIP (contradicts a little cons #1)
> 4. Support many SIP providers.
> 5. Plays nicely with other gnome apps I use.
> 
> 
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-- 
 _      Damien Sandras
(o-     GnomeMeeting: http://www.gnomemeeting.org/
//\     FOSDEM 2005 : http://www.fosdem.org
v_/_    H.323 phone : callto:ils.seconix.com/dsandras seconix com