[OpenSIPS-Users] ms teams ACK

[volga629]

Hello Everyone,

Thank you all for the help.

Here are some highlight

1. Avoid call NAT in reply with 200 OK with option 1 or 3

2. Preset routing header toward Microsoft Teams.

3. Use record route toward you asterisk or feeeswitch.

4. Append contact header in keepalive messages.

5. Use dialplan feature to remove append + to normalize phone numbers.


volga629


On 4/19/20 9:40 AM, volga629 via Users wrote:
>
> Hello Johan,
>
> Thank you for reply.
>
>
> The only NAT problem can be on MS Teams Client,  because on Opensips 
> side pretty sure all good.
>
>
> volga629
>
>
> On 4/19/20 3:04 AM, Johan De Clercq wrote:
>> Can’t it be a NAT problem? The IP address where the bye is coming 
>> from doesn’t seem a pstnhub to me.
>>
>> Outlook voor iOS <https://aka.ms/o0ukef> downloaden
>> ------------------------------------------------------------------------
>> *Van:* Users <users-bounces at lists.opensips.org> namens volga629 via 
>> Users <users at lists.opensips.org>
>> *Verzonden:* Saturday, April 18, 2020 11:01:19 PM
>> *Aan:* OpenSIPS users mailling list <users at lists.opensips.org>; 
>> Alexey Vasilyev <alexei.vasilyev at gmail.com>
>> *Onderwerp:* Re: [OpenSIPS-Users] ms teams ACK
>>
>> Hello Alexey,
>>
>> Thank you on reply,
>>
>> I undone all changes regard headers changes and MS Teams send BYE 
>> directly to asterisk.
>>
>> No Route header present.
>>
>> But INVITE ACK 183 180 all travel with  proper routing information.
>>
>>
>> 2020/04/18 17:54:28.599711 190.109.70.77:5060 -> 190.109.68.250:5060
>> BYE sip:11988582770 at 190.109.68.250:5060 
>> <sip:11988582770 at 190.109.68.250:5060> SIP/2.0
>> FROM: <sip:551120999838 at 190.109.70.77> 
>> <sip:551120999838 at 190.109.70.77>;tag=4d7fb0763c224e39a13a03c669c4b387
>> TO: <sip:11988582770 at 190.109.70.77> 
>> <sip:11988582770 at 190.109.70.77>;tag=as41e97ff5
>> CSEQ: 3 BYE
>> CALL-ID: 2e6c1a8d2383a4752403e94512ced077 at 190.109.70.77 
>> <mailto:2e6c1a8d2383a4752403e94512ced077 at 190.109.70.77>
>> MAX-FORWARDS: 69
>> Via: SIP/2.0/UDP 
>> 190.109.70.77:5060;branch=z9hG4bK050e.e400e373.0;i=66c9c603
>> VIA: SIP/2.0/TLS 
>> 52.114.14.70:5061;rport=8208;received=52.114.14.70;branch=z9hG4bK9594cd7
>> REASON: 
>> Q.850;cause=18;text="fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;Call 
>> Controller timed out while waiting for acknowledgement."
>> CONTACT: 
>> <sip:52.114.14.70:8208;nat=yes;x-i=fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;x-c=b5841f98785c5819bb99e57cd0fa7d86/s/1/9b6ba2f8eefa4a67bed29609fd1884ec> 
>> <sip:52.114.14.70:8208;nat=yes;x-i=fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;x-c=b5841f98785c5819bb99e57cd0fa7d86/s/1/9b6ba2f8eefa4a67bed29609fd1884ec>
>> CONTENT-LENGTH: 0
>> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.13.7 i.ASSE.3
>> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY
>>
>>
>> volga629
>>
>>
>> On 4/18/20 5:13 PM, Alexey Vasilyev wrote:
>>> Hi volga629,
>>>
>>> There were nothing special for ACK. You don't need to change
>>> To/From/Contact. All the necessary steps were in the article
>>> https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/  and for most
>>> people it still works.
>>> So I'm not sure, that MS changed anything, because all the hardware SBCs
>>> should change behaviour, so they need new firmware. SBC vendors should
>>> inform customers to update etc. So this is not so simple process. And it
>>> definitely make no sense for anybody.
>>> And in the test lab for the article I've used absolutely the same
>>> architecture with asterisk, the only difference was RTPEngine to transcode
>>> SRTP-RTP.
>>> And within test lab I've tested not only calls, but transfers worked fine
>>> too.
>>>
>>>
>>>
>>> -----
>>> ---
>>> Alexey Vasilyev
>>> --
>>> Sent from:http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html
>>>
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>>> Users at lists.opensips.org  <mailto:Users at lists.opensips.org>
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
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