If the ptime attribute is present for a stream, it indicates the
desired packetization interval that the offerer would like to
RECEIVE.

Everything you described is how we behave.
We will not be changing it.

Not specifying the ptime is a giant performance hit because we cannot
initilize the timers.
Wanting to send 60 and not actually specifying it could also probably
explained away in some deep interpretation of the RFC but it's not
typical and it's plain foolish.

The only advice I can give you is to create a dedicated sip profile
for this call path and configure the codec negotiation to scrooge and
define G729@60i in your codec config.

This will make FS use 60ms g729 regardless of what it sees in the sdp,
this is not optimal for anyone but your one case which is why I say to
put it in a specific profile.


On Fri, Oct 8, 2010 at 6:57 AM, David Ponzone <david.ponzone@ipeva.fr> wrote:

I am out of my league, I would need to dig into the RFCs so I prefer to wait
for comments from the people who wrote the code, because I am sure they have
an opinion on how the RFCs should be read or why they did what they did.
For the autofix-timing thing, I don't think there is a way to change that
per gateway or with a channel variable.
So yes, you have to use another IP or same IP with non-standard port.
But you just need 2: one profile with it, one without.
David Ponzone  Direction Technique
email: david.ponzone@ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.



Le 08/10/2010 à 13:39, Jan Riedinger a écrit :


Am 08.10.2010 13:06, schrieb David Ponzone:

Jan,
to answer to 2 others questions you ask:
Why FS tries to enforce 20 ?
Well, the default is 20ms for most codecs, except perhaps G723, and no
explicit ptime means 20ms for most codecs.
So your carrier is sending no ptime, meaning they want 20.
FS agrees and send back 20 (and explictly, because smart people always do
things explictly and avoid relying on default values/behaviours).

I think this isn't correct. If you work with a codec list in a Cisco and set
any byte / frame size values for the codecs of the codec list, the Cisco
doesn't specify any ptime in the initial INVITE message, even if for all
codecs of the codec list the same frame size is specified. Thus it's risky
at this point, if FS assums that the caller wants to use 20 ms


So, yes, the message displayed by FS is correct at some point:
FS asked for 20ms, and your carrier is sending 60ms.

But the usage of 60 ms nevertheless, is ok according the RFC.

Now, I see your point: perhaps the phrase is not very clear.
I think the issue is (and Anthony or Brian will correct me on this if
required) that FS tries to negotiate the same ptime on both directions,
because what the RFC says about asymmetrical ptimes is scary, AFAIK. I heard
people reporting major issues trying to do this.

I configured for a long time a payload of 40 or 60 bytes on my Ciscos,
because of the disadvantageous TCP/IP header overhead, if you go with 20
bytes. I asked my business partners to do it in the same way. However, often
the didn't change their standard config and continued to use 20 bytes. I had
trouble by this asymmetry only once out of more than 200 configured
interconnects.

Ok the RFC allows it, but as usual, it was probably badly implemented by
most vendors, and anyway, there is no real benefit.

So FS tries to stay simple.
I think that's what FS means by "We were told": the other party asked us for
20ms, and as we like to keep things simple, we also asked for 20ms, and they
send back 60ms, those p....bast.... :)

As you see in the trace graph attached to my previous e-mail, the re-INVITE
of FFS results in an "internal server error" at the terminating GW. Of
course this shouldn't be the case and doesn't comply with the RFC, but this
problem is caused by the efforts of FS to fix a problem, which doesn' exist
- at least according the RFC.

Basically, I think what you are asking is a new parameter that would
instruct FS to stop trying to re-packetize and accept asymmetrical ptimes.
About the message, you can get rid of it with rtp-autofix-timing=false, but
use it at your own risk.

Is it possible to use rtp-autofix-timing just for a specific carrier? If I
specify it in the default profile, it is used for all carriers.
Maybe/Probably I'm wrong, but according my current knowledge I have to use
another non standard IP port, if I want to use another profile just for this
specific carrier.

BR
    Jan

David Ponzone   Direction Technique
email: david.ponzone@ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34
Service Client  IP eva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com
Ce message et toutes les pièces jointes sont confidentiels et établis à
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
non autorisée est interdite. Tout message électronique est susceptible
d'altération. IPeva décline toute responsabilité au titre de ce message s'il
a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
message, merci de le détruire immédiatement et d'avertir l'expéditeur.



Le 08/10/2010 à 12:40, Jan Riedinger a écrit :

I'm terminating various destination by various carriers. After migrating one
customer to Freeswitch, we observed problems for the termination of a
specific route for a specific carrier. I tried to examine the problem in
detail and I think it's related to problems regarding the ptime negotiation.
I think Freeswitch doesn't breach any RFC, but I'm not sure, if the
behaviour is optimal.

The SDP of the Caller INVITE-Message at time 1160,056 in the attached trace
doesn't include any ptime setting. Nevertheless Freeswitch includes a
ptime=20 media attribute in the forwarded INVITE message at time 1160,065.
The ringing SDP sent by the callee at time 1161,948 again doesn't include
any ptime setting. Nevertheless, Freeswitch includes in the Session Progress
SDP (at time 1164,240) a ptime=20 media atrribute. Why try Freeswitch to
force the usage of ptime=20 for the communication?

The OK SDP at time 1164,240 again doesn't contain a ptime media attribute.
Nevertheless, the Freeswitch add a ptime=20 media attribute forwarded to the
caller at 1164,256.

It seems that the callee is sending in the following with a frame size of 60
bytes - it never claimed to use ptime=20 and according the RFC 3264 it
SHOULD send with ptime=20 because of the received INVITE message
specification, but it DON'T HAVE to send with ptime=20.

At next Freeswitch tries to fix "the issue". In the logfile I found:

e686b430-5d2d-488b-8b58-0fca1965eea7 2010-10-07 15:20:25.673206 [WARNING]
mod_sofia.c:1033 We were told to use ptime 20 but what they meant to say was
60
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken,
who knows what will happen..

This log message isn't correct. The callee never specified anything about
the usage of a specific ptime. Furthermore, according RFC 3264  the ptime
doesn't specify the frame size, which will be used to send packages by the
side, which specify it in the SDP. In the RFC 3264 is written:

If the ptime attribute is present for a stream, it indicates the
desired packetization interval that the offerer would like to
receive .

...

There is now requirement that the packetization interval be the same in
each direction for a particular stream.

IMHO that means, that it isn't possible in principle that a device is lying
about it's ptime usage, because it only specify by the media attribute the
packetization it likes to receive and doesn't specify the packetization it
will use itself.

For fixing "the problem" Freeswitch sends a re-INVITE message at 1164,777.
This message includes in the message header "X-Broken-PTIME: Adv=20;
Sent=60", and ptime = 60 media attribute.
The callee fails to process this re-INVITE and drops the call.

I made the trace after I set the newly introduced parameter
"passthru_ptime_mismatch=true" (it's documented in the Wiki since
yesterday). Does it make sense, that Freeswitch tries to fix any ptime
setting if this variable is set to true?

If someone wants to examine this issue more detailed, I can provide the
Wireshark-cap file of the call and the debug output of Freeswitch.


Thank you in advance
           Jan





--
Jan Riedinger Phone : +49-30-39 73 19 66
Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
E-Mail: riedinger@sns.eu
SNS Consult GmbH ICQ : 163-237-041
Südwestkorso 49a MSN : jan@sns-consult.de
14197 Berlin GERMANY Skype : Jan Riedinger

AG Charlottenburg - HRB 71973

<Ptime Problem Call
Trace.tif>_______________________________________________
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


_______________________________________________
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

--
Jan Riedinger Phone : +49-30-39 73 19 66
Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64
E-Mail: riedinger@sns.eu
SNS Consult GmbH ICQ : 163-237-041
Südwestkorso 49a MSN : jan@sns-consult.de
14197 Berlin GERMANY Skype : Jan Riedinger

AG Charlottenburg - HRB 71973

_______________________________________________
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


_______________________________________________
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