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 <[email protected]> namens volga629 via
Users <[email protected]>
*Verzonden:* Saturday, April 18, 2020 11:01:19 PM
*Aan:* OpenSIPS users mailling list <[email protected]>;
Alexey Vasilyev <[email protected]>
*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:[email protected]:5060
<sip:[email protected]:5060> SIP/2.0
FROM: <sip:[email protected]>
<sip:[email protected]>;tag=4d7fb0763c224e39a13a03c669c4b387
TO: <sip:[email protected]>
<sip:[email protected]>;tag=as41e97ff5
CSEQ: 3 BYE
CALL-ID: [email protected]
<mailto:[email protected]>
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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