Jan,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
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).
But the usage of 60 ms nevertheless, is ok according the RFC.
So, yes, the message displayed by FS is correct at some point:FS asked for 20ms, and your carrier is sending 60ms.
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.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.
Ok the RFC allows it, but as usual, it was probably badly implemented by most vendors, and anyway, there is no real benefit.
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.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.... :)
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.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.
David Ponzone Direction Techniqueemail: david.ponzone@ipeva.frtel: 01 74 03 18 97gsm: 06 66 98 76 34
Service Client IP evatel: 0811 46 26 26
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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.<Ptime Problem Call Trace.tif>_______________________________________________
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 60This 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:
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..
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
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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