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[Freeswitch-users] Calls drop immediately when terminator forces G.729 Codec.

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anthony.minessale at g...
Guest

Posted: Thu Jun 04, 2009 8:00 am Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

you should have also turned in the sip trace
sofia profile internal siptrace on

On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke <jim@evolutiontel.net (jim@evolutiontel.net)> wrote:

Quote:

Hi All,

Looking for some debugging tips and comments on what might be causing
the media port in the 200OK ( Answer message) to be set to 0 by
freeswitch. Essentially it looks like data might be getting trampled
somehow.

Portion of 200OK going into Freeswitch
m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100
NSE/8000..a=
fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=ptime:20..a=sendrecv..

Portion of 200OK coming out of Freeswitch
m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100
NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-15..

Note the media port has been set to 0 and the rtpmap for G729 is also
not correct. On receipt of this bad 200Ok the originator sends a BYE.

We are using FS as a B2BUA with bypass_media set to true. Thus IMHO
Freeswitch should not be touching the SDP portion of the message and
just passing it through.

This can reproduce this at will, so I can collect whatever data is
nessicary. I have added the sofia debug from the console.

Thanks,
--
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@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/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 8:10 am Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

I have traces using ngrep, however if you want to see it all in one
file I will collect tommorow.

Regards,

On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:

Quote:

you should have also turned in the sip trace
sofia profile internal siptrace on

On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke <jim@evolutiontel.net> wrote:

Quote:

Hi All,

Looking for some debugging tips and comments on what might be causing
the media port in the 200OK ( Answer message) to be set to 0 by
freeswitch. Essentially it looks like data might be getting trampled
somehow.

Portion of 200OK going into Freeswitch
m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100
NSE/8000..a=
fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=ptime:20..a=sendrecv..

Portion of 200OK coming out of Freeswitch
m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100
NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-15..

Note the media port has been set to 0 and the rtpmap for G729 is also
not correct. On receipt of this bad 200Ok the originator sends a BYE.

We are using FS as a B2BUA with bypass_media set to true. Thus IMHO
Freeswitch should not be touching the SDP portion of the message and
just passing it through.

This can reproduce this at will, so I can collect whatever data is
nessicary. I have added the sofia debug from the console.

Thanks,
--
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@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/

AIM: anthm
MSN:anthony_minessale@hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org
pstn:213-799-1400

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

--
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 6:15 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Hi Anthony,

Traces as requested. Let me know if you want a jira opened or any further data.

Regards,
Jim

On Thu, Jun 4, 2009 at 10:59 PM, Anthony Minessale
<anthony.minessale@gmail.com> wrote:

Quote:

you should have also turned in the sip trace
sofia profile internal siptrace on

On Thu, Jun 4, 2009 at 7:43 AM, Jim Burke <jim@evolutiontel.net> wrote:

Quote:

Hi All,

Looking for some debugging tips and comments on what might be causing
the media port in the 200OK ( Answer message) to be set to 0 by
freeswitch. Essentially it looks like data might be getting trampled
somehow.

Portion of 200OK going into Freeswitch
m=audio 16416 RTP/AVP 18 100 101..a=rtpmap:18 G729a/8000..a=rtpmap:100
NSE/8000..a=
fmtp:100 192-193..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=ptime:20..a=sendrecv..

Portion of 200OK coming out of Freeswitch
m=audio 0 RTP/AVP 96 100 101..a=rtpmap:96 G729a/8000..a=rtpmap:100
NSE/8000..a=fmtp:100 192-193..a=rtpmap:101 telephone-event
/8000..a=fmtp:101 0-15..

Note the media port has been set to 0 and the rtpmap for G729 is also
not correct. On receipt of this bad 200Ok the originator sends a BYE.

We are using FS as a B2BUA with bypass_media set to true. Thus IMHO
Freeswitch should not be touching the SDP portion of the message and
just passing it through.

This can reproduce this at will, so I can collect whatever data is
nessicary. I have added the sofia debug from the console.

Thanks,
--
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Thu Jun 04, 2009 6:30 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Port 0 indicates a rejection. Something is sending a 200 OK with 0 and g729a on codec 96. Have you modified the freeswitch code ?

/b

On Jun 4, 2009, at 6:15 PM, Jim Burke wrote:

Quote:

Hi Anthony,

Traces as requested. Let me know if you want a jira opened or any further data.

Regards,
Jim

Brian West
brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 6:50 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Using FreeSWITCH Version 1.0.trunk (13523) and have not modified the code.

Yes, exactly this is what causes the originator to release the call.
As you will see in the traces, the 200OK is good on the way into FS,
but looks to be trampled on the way out Sad

Regards,

On Fri, Jun 5, 2009 at 9:28 AM, Brian West <brian@freeswitch.org> wrote:

Quote:

Port 0 indicates a rejection. Something is sending a 200 OK with 0 and
g729a on codec 96. Have you modified the freeswitch code ?
/b
On Jun 4, 2009, at 6:15 PM, Jim Burke wrote:

Hi Anthony,

Traces as requested. Let me know if you want a jira opened or any further
data.

Regards,
Jim

Brian West
brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

brian at freeswitch.org
Guest

Posted: Thu Jun 04, 2009 6:52 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

But that SDP is not generated by FreeSWITCH... are you using proxy media or something? maybe bypass?

/b

On Jun 4, 2009, at 6:49 PM, Jim Burke wrote:

Quote:

Yes, exactly this is what causes the originator to release the call.
As you will see in the traces, the 200OK is good on the way into FS,
but looks to be trampled on the way out Sad

Regards,

Brian West
brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 7:08 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

bypass_media is set, and proxy_media is not set.

I agree, FS should not be touching the SDP for calls in bypass_media
mode. Interesting, when you look in the file when FS reports the
Remote SDP it still looks ok, then a little further down you can see
it send the 200OK out to the originator and that SDP in-correctly
reports the media port.

The other interesting point is a=rtpmap:96 G729a/8000. 96 is not the
correct rtpmap for G729 and it is not mentioned on the incoming 200Ok
to FS

Regards,

On Fri, Jun 5, 2009 at 9:51 AM, Brian West <brian@freeswitch.org> wrote:

Quote:

But that SDP is not generated by FreeSWITCH... are you using proxy media or
something? maybe bypass?
/b
On Jun 4, 2009, at 6:49 PM, Jim Burke wrote:

Yes, exactly this is what causes the originator to release the call.
As you will see in the traces, the 200OK is good on the way into FS,
but looks to be trampled on the way out Sad

Regards,

Brian West
brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 7:25 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Hey Brian

Quote:

From your comments above this appears to be the code that does the

damage. I guess now the question is why??

soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called
soa_static(0x9c1d4e8, soa_generate_answer): generating local description
soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description
soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media

Regards,

On Fri, Jun 5, 2009 at 10:07 AM, Jim Burke <jim@evolutiontel.net> wrote:

Quote:

--
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net

brian at freeswitch.org
Guest

Posted: Thu Jun 04, 2009 7:35 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Try SVN trunk I cna tell you're using older code! Wink

/b

On Jun 4, 2009, at 7:23 PM, Jim Burke wrote:

Quote:

Hey Brian

Quote:

From your comments above this appears to be the code that does the

Brian West
brian@freeswitch.org (brian@freeswitch.org)

-- Meet us at ClueCon! http://www.cluecon.com

jim at evolutiontel.net
Guest

Posted: Thu Jun 04, 2009 8:01 pm Post subject: [Freeswitch-users] Calls drop immediately when terminator fo

Hmmm...no luck with the SVN trunk FreeSWITCH Version 1.0.trunk (13624)

On Fri, Jun 5, 2009 at 10:34 AM, Brian West <brian@freeswitch.org> wrote:

Quote:

Try SVN trunk I cna tell you're using older code! Wink

/b
On Jun 4, 2009, at 7:23 PM, Jim Burke wrote:

Hey Brian

From your comments above this appears to be the code that does the

damage. I guess now the question is why??

soa_static_offer_answer_action(0x9c1d4e8, soa_generate_answer): called
soa_static(0x9c1d4e8, soa_generate_answer): generating local description
soa_static(0x9c1d4e8, soa_generate_answer): upgrade with remote description
soa_static(0x9c1d4e8, soa_generate_answer): marking rejected media

Regards,

Brian West
brian@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com

_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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