AW: [Linphone-users] AW: linphone-1.0.0 is out / playing audio when receiving sdp/status 183 etc.

[David Schumacher]

hello,

i attached a log from a call to a wrong mobile-number where linphone stays
silent (kphone for example plays the informational message from the
phone-company in this case).

the linphone-machine has ip 192.168.1.212, the remote-side (running
asterisk) has ip 192.168.1.217.

thanks for your time!

dave

p.s.: at line 8 of the attached log is an entry like: "| INFO1 | <eXutils.c:
416> Outgoing interface to reach 15.128.128.93 is 192.168.1.212.". whats up
with that ip "15.128.128.93"?


-----Ursprüngliche Nachricht-----
Von: Simon Morlat [mailto:address@hidden
Gesendet: Dienstag, 12. April 2005 17:30
An: address@hidden
Cc: David Schumacher
Betreff: Re: [Linphone-users] AW: linphone-1.0.0 is out / playing audio
when receiving sdp/status 183 etc.


Hello,

I don't think it's an eXosip problem. Seems to be a pure audiostreaming
problem but I can't see why. I'll have a look at the code again, otherwise
you can help me by sending me a complete log from a 183'd call, use
linphonec
-d 6 or linphone --verbose .
Thanks
Simon

Le Mercredi 23 Mars 2005 11:53, David Schumacher a écrit :
> hello,
>
> congratulations to the milestone-release ;-)
>
> i tried again to figure out how to make linphone start the audiostream
when
> a sdp with other status than 200 is reveiced by linphone.
>
> exosip seems to return "call ringing" (EXOSIP_CALL_RINGING) when the
remote
> side sends a status 183 with sdp, and ethereal shows a lot of udp-traffic
> on port 7078 (audio-data?), but linphone stays silent, no local ring or
> other sound. when the remote side gives status 180, local ring is played.
> the voice-mediastream is only opened when the remote side sends status
200.
> probably an issue with exosip?
>
> i saw that you changed the handling of sdp-messages in pre10
> (coreapi/osipuacb.c, linphone_call_ringing() ). that suppresses local
> ringing, but the voice-audio (mostly informational messages) is not
played.
>
> i also tried kphone and sjphone, they both play all audio-data they
receive
> from the remote side ;-), but they are both not scriptable ;-(, so i will
> continue to use linphonec, controlled by a small pure-perl http-server,
> dialing and answering ist done in an iframe (obviously loaded from
> http://localhost/...) embedded in a intranet-webpage.
>
> greetings from germany,
> dave
>
> p.s.: i dont use any proxy/registrar-functionality, i just dial
> "sip:address@hidden", if this makes any differences in
> call-handling.
>
>
> -----Ursprungliche Nachricht-----
> Von: address@hidden
> [mailto:address@hidden Auftrag von
> Simon Morlat
> Gesendet: Dienstag, 22. Marz 2005 12:18
> An: address@hidden
> Cc: address@hidden
> Betreff: [Linphone-users] linphone-1.0.0 is out.
>
>
> Hello,
>
> After more than 1 year of silence and 13 pre-releases linphone-1.0.0 comes
> out !
> Download the source at:
> http://simon.morlat.free.fr/download/1.0.0/source
>
> Some official debian-sid packages will come soon, rpms are welcome.
>
> No more features since pre13, but a lot of things since 0.12.2, see the
> ChangeLog file for details.
>
> Some known problems:
>  - it seems ipv6 does not work very well, help is welcome.
>  - the html manual does not display in gnome help browser, and is
> completely outdated. Help is welcome, of course.
>
>
> I hope to make more frequent releases since now. The gap from 0.12.2 to
> 1.0.0
> was difficult because of a radical change in the sip signaling part (move
> from osipua library to eXosip ).
>
>
>
> Simon
>
>
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>
>
>
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linphonelog_183.txt
Description: Text document