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[nico at vthadden.de]

Hi,

I'm currently investigating reproducable call drops when calling
mobile-phone numbers from Deutsche Telekom (T-Mobile Germany).
My provider has a NGN (SIP) Interconnection with Deutsche Telekom and is
transparend for the messages involved here (although IP rewriting is
happening).

The calls contain following SDP in the invite from my FreeSwitch towards
my provider:
v=0                                                                                                                                                                                                                                                                                                                         

o=FreeSWITCH 1608967203 1608967204 IN IP4 *redacted*
                                                                                                                                                                                                                                                                    

s=FreeSWITCH                                                                                                                                                                                                                                                                                                                

c=IN IP4  *redacted*
                                                                                                                                                                                                                                                                                                  

t=0
0                                                                                                                                                                                                                                                                                                                       

m=audio 32256 RTP/AVP 9 8 0
101                                                                                                                                                                                                                                                                                             

a=rtpmap:9
G722/8000                                                                                                                                                                                                                                                                                                        

a=rtpmap:8
PCMA/8000                                                                                                                                                                                                                                                                                                        

a=rtpmap:0
PCMU/8000                                                                                                                                                                                                                                                                                                        

a=rtpmap:101
telephone-event/8000                                                                                                                                                                                                                                                                                           

a=fmtp:101
0-16                                                                                                                                                                                                                                                                                                             

a=ptime:20  

Once the call is established, FS sends a re-invite after 50% of the
expiration timer is elapsed, 15 minutes in this case. The re-invite
contains a slightly modified SDP:
v=0                                                                                                                                                                                                                                                                                                                         

o=FreeSWITCH 1608967203 1608967204 IN IP4
*redacted*                                                                                                                                                                                                                                                                     

s=FreeSWITCH                                                                                                                                                                                                                                                                                                                

c=IN IP4
*redacted*                                                                                                                                                                                                                                                                                                      

t=0
0                                                                                                                                                                                                                                                                                                                       

m=audio 32256 RTP/AVP 8 101 9
0                                                                                                                                                                                                                                                                                             

a=rtpmap:8
PCMA/8000                                                                                                                                                                                                                                                                                                        

a=rtpmap:101
telephone-event/8000                                                                                                                                                                                                                                                                                           

a=fmtp:101
0-16                                                                                                                                                                                                                                                                                                             

a=rtpmap:9
G722/8000                                                                                                                                                                                                                                                                                                        

a=rtpmap:0
PCMU/8000                                                                                                                                                                                                                                                                                                        

a=ptime:20   

The new codec on position 1 (PCMA) is not necessary the chosen one for
the session, that call was using G.722 (verified via 'show channels').

Telekom does not like my SDP change and responds with:
SIP/2.0 488 SDP Parameter Error In SIP
Request                                                                                                                                                                                                                                                                              


The freeswitch console only logs:
2020-12-26 16:32:49.578925 [DEBUG] sofia.c:7326 Channel
sofia/external/*redacted*entering state [calling][0]
2020-12-26 16:32:49.618925 [DEBUG] sofia.c:7319 Channel
sofia/external/*redacted* skipping state [ready][488]

The call is disconnected 15 minutes later because the session timer has
expired:
2020-12-26 16:47:53.838925 [NOTICE] sofia.c:1089 Hangup
sofia/external/*redacted* [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]

This does not happen when I only allow PCMA (or any other codec) since
the SDP can't get mixed up.

I have reproduced that behaviour with
1.10.5~release~6~25569c1631~buster-1~buster+1 and
1.6.20~37~987c9b9-1~jessie+1.

Do you have any idea why FS changes the SDP (without reason?) and what I
can do about it?

Thanks
Nico

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[brians at iptel.co]

Unless you're doing hd audio on these calls your solution is to just use g711 on calls into this provider.

On Saturday, December 26, 2020, Nicola von Thadden <nico@vthadden.de (nico@vthadden.de)> wrote:

[nico at vthadden.de]

Hi,

yes, we are using HD audio there. Deutsche Telekom is codec transparent, they even pass opus and other codecs since a while.
So disabling any codecs but G.711 is not really a solution, especially since DT is the biggest provider in Germany and that issue might not only affect my setup but in theory all other Freeswitch in Germany here which might want to call a DT landline or mobile number.

Is there a way to make freeswitch stop mixing up the SDP?

Nico

On 12/28/20 12:04 AM, Brian : wrote:

[brian at freeswitch.com]

I've seen this problem before: the remote end is preferring PCMA over PCMU when it does the session timer, If you're talking to any providers in the EU, disallow PMCU always.

Can you provide a full sip trace of this?


On Mon, Dec 28, 2020 at 9:16 AM Nicola von Thadden <nico@vthadden.de (nico@vthadden.de)> wrote: