webstean
Customer
- Joined
- Jul 14, 2021
- Messages
- 11
- Reaction score
- 1
Hi All,
I've setup a kamailio (kamailio.org) as a registered SBC for Microsoft Direct Routing proxying into a 3CX instance. Both kamailio and 3CX are in the same private network (VNET) in Azure with acces to the Internet via NAT. 3CX was built with express.
So I can receive phone calls from Microsoft Teams in 3CX - which is great, except the RTP media setup fails, so the call it never answered.
So first questioin, is 3CX actually compatible with Microsoft Teams codecs - or do you need to transcode?
FYI: As the 3CX instance is in the cloud, I have 3CX SBCs at each site as Raspberry Pis.
BTW: All Direct Routing Microsoft Teams calls are encrypted RTP/SAVP and I have media bypass TURN OFF (the default).
I believe the firewall rules are correct - so that shouldn't be the issue
Some reference material:
https://www.3cx.com/docs/sip-trunk-codecs-sdp/
https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan
I'm having trouble understanding the above.
Any help suggestions/appreciated
Here's a sample invite:-
07/14/2021 11:06:17 AM - [CM500002]: Call(C:340): Info on incoming INVITE from Line:10001<<anonymous:
Invite-IN Recv Req INVITE from 10.0.0.6:58320 tid=6dfe.472527ab836ad1dad51c7dd997839cfc.0 Call-ID=6a99ca46cd2559eca7958b167f24f6c0:
INVITE sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 52.189.194.44:5061;branch=z9hG4bK6dfe.472527ab836ad1dad51c7dd997839cfc.0;received=10.0.0.6;i=f
Via: SIP/2.0/TLS 52.114.148.0:5061;rport=8513;branch=z9hG4bKf7244afb
Max-Forwards: 69
Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
Contact: <sip:api-du-c-auea.pstnhub.microsoft.com:443;x-i=53745319-775e-414f-a9f5-1da8b075b80e;x-c=6a99ca46cd2559eca7958b167f24f6c0/d/10/ba8234ca8f7f4aa091cb7c2c2b8f9a89>
To: <sip:[email protected]:5061;user=phone>
From: <sip:[email protected]>;tag=bb1454cf156d4b2b8923abbf5b03d699
Call-ID: 6a99ca46cd2559eca7958b167f24f6c0
CSeq: 1 INVITE
Session-Expires: 3600
Min-SE: 300
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY
Content-Type: application/sdp
Supported: timer
User-Agent: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.USWE2.7
Privacy: id
P-Asserted-Identity: <sip:[email protected]>
Content-Length: 1105
v=0
o=- 140019 0 IN IP4 127.0.0.1
s=session
c=IN IP4 52.114.192.162
b=CT:10000000
t=0 0
m=audio 52872 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118
c=IN IP4 52.114.192.162
a=rtcp:52873
a=ice-ufrag:XFX6
a=ice-pwd:ImzOTcSdXPjEAhwX+ZJiiAWr
a=rtcp-mux
a=candidate:1 1 UDP 2130706431 52.114.192.162 52872 typ srflx raddr 10.0.33.52 rport 52872
a=candidate:1 2 UDP 2130705918 52.114.192.162 52873 typ srflx raddr 10.0.33.52 rport 52873
a=candidate:2 1 tcp-act 2121006078 52.114.192.162 49152 typ srflx raddr 10.0.33.52 rport 49152
a=candidate:2 2 tcp-act 2121006078 52.114.192.162 49152 typ srflx raddr 10.0.33.52 rport 49152
a=label:main-audio
a=mid:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RaqXOiKePBlncGJ5KzzcmIKtxU9zym7rfkkQINpc|2^31
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20
I've setup a kamailio (kamailio.org) as a registered SBC for Microsoft Direct Routing proxying into a 3CX instance. Both kamailio and 3CX are in the same private network (VNET) in Azure with acces to the Internet via NAT. 3CX was built with express.
So I can receive phone calls from Microsoft Teams in 3CX - which is great, except the RTP media setup fails, so the call it never answered.
So first questioin, is 3CX actually compatible with Microsoft Teams codecs - or do you need to transcode?
FYI: As the 3CX instance is in the cloud, I have 3CX SBCs at each site as Raspberry Pis.
BTW: All Direct Routing Microsoft Teams calls are encrypted RTP/SAVP and I have media bypass TURN OFF (the default).
I believe the firewall rules are correct - so that shouldn't be the issue
Some reference material:
https://www.3cx.com/docs/sip-trunk-codecs-sdp/
https://docs.microsoft.com/en-us/microsoftteams/direct-routing-plan
I'm having trouble understanding the above.
Any help suggestions/appreciated
Here's a sample invite:-
07/14/2021 11:06:17 AM - [CM500002]: Call(C:340): Info on incoming INVITE from Line:10001<<anonymous:
Invite-IN Recv Req INVITE from 10.0.0.6:58320 tid=6dfe.472527ab836ad1dad51c7dd997839cfc.0 Call-ID=6a99ca46cd2559eca7958b167f24f6c0:
INVITE sip:[email protected]:5061;user=phone;transport=tls SIP/2.0
Via: SIP/2.0/TLS 52.189.194.44:5061;branch=z9hG4bK6dfe.472527ab836ad1dad51c7dd997839cfc.0;received=10.0.0.6;i=f
Via: SIP/2.0/TLS 52.114.148.0:5061;rport=8513;branch=z9hG4bKf7244afb
Max-Forwards: 69
Record-Route: <sip:sip-du-a-us.pstnhub.microsoft.com:5061;transport=tls;lr>
Contact: <sip:api-du-c-auea.pstnhub.microsoft.com:443;x-i=53745319-775e-414f-a9f5-1da8b075b80e;x-c=6a99ca46cd2559eca7958b167f24f6c0/d/10/ba8234ca8f7f4aa091cb7c2c2b8f9a89>
To: <sip:[email protected]:5061;user=phone>
From: <sip:[email protected]>;tag=bb1454cf156d4b2b8923abbf5b03d699
Call-ID: 6a99ca46cd2559eca7958b167f24f6c0
CSeq: 1 INVITE
Session-Expires: 3600
Min-SE: 300
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, NOTIFY
Content-Type: application/sdp
Supported: timer
User-Agent: Microsoft.PSTNHub.SIPProxy v.2021.6.15.17 i.USWE2.7
Privacy: id
P-Asserted-Identity: <sip:[email protected]>
Content-Length: 1105
v=0
o=- 140019 0 IN IP4 127.0.0.1
s=session
c=IN IP4 52.114.192.162
b=CT:10000000
t=0 0
m=audio 52872 RTP/SAVP 104 9 103 111 18 0 8 97 101 13 118
c=IN IP4 52.114.192.162
a=rtcp:52873
a=ice-ufrag:XFX6
a=ice-pwd:ImzOTcSdXPjEAhwX+ZJiiAWr
a=rtcp-mux
a=candidate:1 1 UDP 2130706431 52.114.192.162 52872 typ srflx raddr 10.0.33.52 rport 52872
a=candidate:1 2 UDP 2130705918 52.114.192.162 52873 typ srflx raddr 10.0.33.52 rport 52873
a=candidate:2 1 tcp-act 2121006078 52.114.192.162 49152 typ srflx raddr 10.0.33.52 rport 49152
a=candidate:2 2 tcp-act 2121006078 52.114.192.162 49152 typ srflx raddr 10.0.33.52 rport 49152
a=label:main-audio
a=mid:1
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:RaqXOiKePBlncGJ5KzzcmIKtxU9zym7rfkkQINpc|2^31
a=sendrecv
a=rtpmap:104 SILK/16000
a=rtpmap:9 G722/8000
a=rtpmap:103 SILK/8000
a=rtpmap:111 SIREN/16000
a=fmtp:111 bitrate=16000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 RED/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=rtpmap:118 CN/16000
a=ptime:20