[Freeswitch-users] Loss of first second of media

[Fraser Redmond]

Thanks Anthony,

Finally managed to get a sip trace - could you do me a favor and take a look
and/or give me some ideas of what to look for?

http://pastebin.freeswitch.org/14300

I've highlighted lines 168 and 193. In between these lines is where the
number is dialed and rings once, then picks up, then theres silence for a
second or two, and that second SIP message is when I start hearing audio.

Thanks,
Fraser




On Thu, Oct 21, 2010 at 6:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> its a blue message on cli
>
> It could also be the other side expecting us to send media first or
> something silly.
> try getting a sip trace and figure out when the rtp starts arriving.
>
>
> On Thu, Oct 21, 2010 at 12:32 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
> > Sorry, yes, I am setting ignore_early_media=true in the first area. (Or
> are
> > you saying that should be off? I forget now why I needed it on, but there
> > was a reason I added it.)
> >
> > Yes, the bridge doesn't start until after the A-leg has answered.
> >
> > Thanks for the suggestion about nat/auto-changing port, I'll have a look
> > into that - would that be in the cli output or in a sip trace? I've
> already
> > looked and it's not appearing in the CLI output (with loglevel=debug),
> > haven't looked in the sip trace yet.
> >
> > Cheers,
> > Fraser
> >
> >
> >
> >
> > On Thu, Oct 21, 2010 at 6:13 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> are you setting ignore_early_media=true in the first vars=values area?
> >>
> >> This looks like you could be calling one leg who is still not answered
> >> and then bridging it to another dest.  The bridge app will wait for
> >> the first leg to answer before bridging.
> >>
> >> Also if you have any NAT anywhere, look for an "auto-changing port"
> >> type message which can also be attributed to this due to a detection
> >> period for incorrect ports.
> >>
> >>
> >>
> >> On Thu, Oct 21, 2010 at 12:00 PM, Fraser Redmond
> >> <fraserredmond at gmail.com> wrote:
> >> > event_socket:
> >> > api originate {vars=values}user/$fromExtn at Domain
> >> > 'set:bLegVars=values,transfer:$toNum xml outbound_call' inline
> >> >
> >> > then
> >> >
> >> > dialplan:
> >> > <action application="set"
> >> > data="effective_caller_id_number=+1800number"/>
> >> > <action application="set" data="effective_caller_id_name="/>
> >> > (set and/or export a bunch of other vars too)
> >> > <action application="set"
> >> >
> >> > data="dial_string=sofia/gateway/
> gatewayname.com/00${destination_number}<http://gatewayname.com/00$%7Bdestination_number%7D>
> "/>
> >> > <action application="bridge" data="${dial_string}"/>
> >> >
> >> > Cheers,
> >> > Fraser
> >> >
> >> >
> >> >
> >> >
> >> > On Thu, Oct 21, 2010 at 5:35 PM, Anthony Minessale
> >> > <anthony.minessale at gmail.com> wrote:
> >> >>
> >> >> how are you accomplishing that? by which technique?
> >> >>
> >> >> On Thu, Oct 21, 2010 at 11:12 AM, Fraser Redmond
> >> >> <fraserredmond at gmail.com> wrote:
> >> >> > The call is originated from Freeswitch (via CLI) to a softphone,
> then
> >> >> > when
> >> >> > that is connected it bridges out to the gateway.
> >> >> >
> >> >> > Cheers,
> >> >> > Fraser
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > On Thu, Oct 21, 2010 at 4:28 PM, Anthony Minessale
> >> >> > <anthony.minessale at gmail.com> wrote:
> >> >> >>
> >> >> >> Where is the other side of this call coming from?
> >> >> >>
> >> >> >> [ (   ) ] -> FS -> (PSTN via SIP)
> >> >> >>
> >> >> >> What goes in the empty space above?
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:+19193869900
> >>
> >> _______________________________________________
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> >
> >
> > _______________________________________________
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> >
> >
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
> _______________________________________________
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