[Freeswitch-users] No audio after Remote SDP:
Anthony Minessale
anthony.minessale at gmail.com
Wed Dec 30 06:04:39 PST 2009
are you using more than one profile here? if so can you repeat the trace
with siptrace on in both profiles.
I notice this on the trace:
1. Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-ee832ee7
2. From: 1000 <sip:1000 at 192.168.1.120>;tag=70c4b62bd9443c9fo0
and this:
sofia_glue.c:2344 AUDIO RTP [sofia/internal/1000 at 192.168.1.120] 192.168.1
.120 port 28490 -> 192.168.1.121 port 16464 codec: 2 ms: 20
Are you behind double nat and or is FS also behind nat?
To address the gentleman who mentioned he was spoiled by asterisk. Keep in
mind we have several advanced nat techniques but many of them break other
situations and they are not all turned on at once with one parameter like in
asterisk, we prefer you use the appropriate ones based on the situation.
On Wed, Dec 30, 2009 at 3:42 AM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:
> Hi Brian,
>
> But that is all I see... I don't see any further messages after the
> 401 (when I have 'NAT Mapping Enabled' selected) ..... I realise that
> this is part of the authentication procedure, but it stops at the
> 401/Authorized message
>
> What about in the pastebin? Do you see anything there? This is when
> the call gets setup, but the IP address in the SDP says a private IP
> address http://pastebin.freeswitch.org/11632
>
> On Tue, Dec 29, 2009 at 2:25 AM, Brian West <brian at freeswitch.org> wrote:
> > If you're using the 401 as an indication that it fails then you don't
> understand how digest authentication works. I would have to see what
> happens after the 401 to see if it really did fail.
> >
> > /b
> >
> > On Dec 24, 2009, at 5:16 AM, Mark Campbell-Smith wrote:
> >
> >> This is all I see and then registration fails.
> >
> >
> > _______________________________________________
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> >
>
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--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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