[Freeswitch-users] Loss of first second of media

Anthony Minessale anthony.minessale at gmail.com
Fri Oct 29 13:39:09 PDT 2010 

OK so,
The phone sends a 180 ringing with NO SDP
then it starts sending RTP
That's is not right.  It's a bug in the phone.



On Fri, Oct 29, 2010 at 1:35 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> can you try another one with just udp and not "port 5060"
> so I can see the rtp too
>
>
> On Fri, Oct 29, 2010 at 12:28 PM, Fraser Redmond
> <fraserredmond at gmail.com> wrote:
>> Thanks Anthony, it's here:
>> http://pastebin.freeswitch.org/14350
>>
>> And pcap is attached.
>>
>> The call connects around (or just before) the 16:58:35 mark (line 558 is
>> what I see in the terminal while waiting for it to connect - both
>> early-media and the missing start of the media)
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>> On Thu, Oct 28, 2010 at 3:59 PM, Anthony Minessale
>> <anthony.minessale at gmail.com> wrote:
>>>
>>> Can you do this trace with debug level logging in addition to the sip
>>> trace
>>> console loglevel debug
>>>
>>> you also may want to get a pcap of it
>>>
>>> tshark udp and port 5060 -w test.pcap
>>>
>>>
>>>
>>> On Sat, Oct 23, 2010 at 11:44 AM, Fraser Redmond
>>> <fraserredmond at gmail.com> wrote:
>>> > 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${destination_number}"/>
>>> >> >> > <action application="bridge" data="${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
>>> >> >> >
>>> >> >> >
>>> >> >> > UNSUBSCRIBE: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
>>> >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> >> IRC: irc.freenode.net #freeswitch
>>> >> >>
>>> >> >> FreeSWITCH Developer Conference
>>> >> >> sip:888 at conference.freeswitch.org
>>> >> >> googletalk:conf+888 at conference.freeswitch.org
>>> >> >> pstn:+19193869900
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >>
>>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE: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
>>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> >> IRC: irc.freenode.net #freeswitch
>>> >>
>>> >> FreeSWITCH Developer Conference
>>> >> sip:888 at conference.freeswitch.org
>>> >> googletalk:conf+888 at conference.freeswitch.org
>>> >> pstn:+19193869900
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE: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
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org
>>> googletalk:conf+888 at conference.freeswitch.org
>>> pstn:+19193869900
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE: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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>



-- 
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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900

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