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CM Register over SIP Trunk

Hi guys,

would it be possible to allow sip users to register over the sip trunk on the Call Manager? or is this method not allowed?

Thanks.

Best regards

 

14 REPLIES
Silver

You're not using SIP trunk

You're not using SIP trunk for registering users - you're using it for call routing. You register SIP devices as official one or third-party on CM.

On other hand if you want to register SIP for PSTN access you would need CUBE gate for that.

HTH,

Dragan

HTH, Dragan
Hall of Fame Super Silver

What are the device types the

What are the device types the users use? As correctly pointed out by Dragan the devices dont register via SIP trunk, but they can be SIP devices and register with CUCM as SIP endpoint.

 

Chris

New Member

Hi Chris,thanks for the reply

Hi Chris,

thanks for the reply.

well the devices are sipml5/html5 sip clients. I'm trying the webrtc with the CM in my lab to see how it works.

The problem is that when I call from a sip client to a phone registered on the CM I've got: SIP/2.0 401 Unauthorized, so the call fails. I thought this might be due to the fact that the CM doesn't know the source IP of the INVITE. CM has no sip trunks.

If I set up a trunk on the CM to the webrtc gateway, then the sip clients try to register over the trunk and then I've got:

RECV:SIP/2.0 405 Method Not Allowed
Warning: 399 cucm "SIP trunk disallows REGISTER"

Any ideas? Do you have any experience with webrtc?

Thanks.

Hall of Fame Super Silver

What are the phones defined

What are the phones defined as in CUCM? remember not all SIP phones are offically supported by Cisco, in fact only handful of them are. Is it defined as 3rd party SIP device?

Chris

New Member

Third-party SIP Device. the

Third-party SIP Device. the phones do register on cm. (if the trunk is not set)

Thanks

Hi Cristi ,

Hi Cristi ,

Can you please send across sip messages. And which protocol are you using as WEBrtc communication protocol ?

See if one end of call leg is Web and other end is SIP , then you need intelligent WEBrtc server to interwork with these two protocols.

 

Thanks

Manish

New Member

Hi Manish,between sip client

Hi Manish,

between sip client and webrtc gw -> ws and between webrtc gw and CM -> sip.

here are the sip messages.

both phones are registered, 9000 is a 7912 and 8080 is sip.

192.168.15.2 - CM

192.168.15.202 - webrtc

SEND: INVITE sip:9000@192.168.15.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:8080@192.168.15.2>;tag=400660433
To: <sip:9000@192.168.15.2>
Contact: <sip:8080@192.168.15.202:10060;ws-src-ip=192.168.251.105;ws-src-port=50731;ws-src-proto=ws;transport=udp>
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Content-Type: application/sdp
Content-Length: 978
Max-Forwards: 70
Authorization: Digest username="8080",realm="ccmsipline",nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6",uri="sip:9000@192.168.15.2",response="352cb2e17e36b32ee4e0d52443d0a106",algorithm=MD5
User-Agent: webrtc2sip Media Server 2.6.0

v=0
o=doubango 1983 678901 IN IP4 192.168.15.202
s=-
c=IN IP4 192.168.15.202
t=0 0
a=tcap:1 RTP/SAVPF RTP/SAVP RTP/AVPF
m=audio 58690 RTP/AVP 8 0 101
c=IN IP4 192.168.15.202
a=ptime:20
a=minptime:1
a=maxptime:255
a=silenceSupp:off - - - -
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=acap:1 crypto:1 AES_CM_128_HMAC_SHA1_80 inline:1YfBfgbhIdMB6YVtyZgJqc77QPHwm9o42aEPbkHD
a=acap:2 crypto:2 AES_CM_128_HMAC_SHA1_32 inline:fujGVOi70hQnKkeUimcFUw2bH3ajZ2iW0xKy5Nrw
a=pcfg:1 t=1 a=1,2
a=pcfg:2 t=2 a=1,2
a=pcfg:3 t=3
a=sendrecv
a=rtcp-mux
a=ssrc:4034073057 cname:c08c56217e96dbc1e8234373eb5d2fcc
a=ssrc:4034073057 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:4034073057 label:doubango@audio
a=ice-ufrag:uaektHZ6KFVn1fw
a=ice-pwd:HAj21nuOrDmIKl3ANXTc3K
a=candidate:tWR5PLw1x 1 udp 2130706431 192.168.15.202 58690 typ host
a=candidate:tWR5PLw1x 2 udp 2130706430 192.168.15.202 58691 typ host


RECV:SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:8080@192.168.15.2>;tag=400660433
To: <sip:9000@192.168.15.2>
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
Content-Length: 0


RECV:SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:8080@192.168.15.2>;tag=400660433
To: <sip:9000@192.168.15.2>;tag=856401750
Date: Wed, 19 Mar 2014 13:26:05 GMT
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 INVITE
Allow-Events: presence
WWW-Authenticate: Digest realm="ccmsipline", nonce="gHqGqDWK4zTzv6Ijl6ixW58AK/Gm4yC6", algorithm=MD5
Content-Length: 0


SEND: ACK sip:9000@192.168.15.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.15.202:10060;branch=z9hG4bK-1536671115;rport
From: <sip:8080@192.168.15.2>;tag=400660433
To: <sip:9000@192.168.15.2>;tag=856401750
Call-ID: df093113-7c32-2a2f-2372-8af2dfbe9235
CSeq: 1875466830 ACK
Content-Length: 0
Max-Forwards: 70

Receiving SIP o/ WebSocket message: ACK sip:9000@192.168.15.2 SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKZEk81zTwfVde8oImts6ZHiTzchfBWh1N;rport
From: "8080"<sip:8080@192.168.15.2>;tag=XFKqC4zu0S9QfzzMzQ4u
To: <sip:9000@192.168.15.2>;tag=1464334432
Call-ID: ecc84fa2-3de3-d953-527f-5e7515cabca3
CSeq: 29519 ACK
Content-Length: 0
Route: <sip:192.168.15.2:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70


Thanks.

Really sorry for delay in

Really sorry for delay in reply.

On CUCM  device- > phone page , what ip address do you see next to sip client which is registered as third party sip device ?

Is it same as webrtc gw ip 192.168.15.202 ?

 

https://supportforums.cisco.com/blog/12016616/webrtc-demo-system

 

Thanks

Manish

New Member

yes, it's the same as webrtc

yes, it's the same as webrtc - 192.168.15.202.

any ideas?

Thanks.

What happens when you call

What happens when you call from cisco phone to sip client ? Does it connect ?

And when you call from sipml client to cisco phone what message do you hear when call gets disconnected ?

 

Thanks

Manish

New Member

it doesn't really connect, it

it doesn't really connect, it rings and when I answer the call gets terminated on the sip client side but the cisco phone stays "connected", as if it is connected. on webrtc console I got this, which is a bit strange because I only enbaled g711. so I believe something's wrong on webrtc side when I call from cisco -> sip.

***ERROR: function: "tdav_session_audio_start()"
file: "src/audio/tdav_session_audio.c"
line: "389"
MSG: No codec matched
***ERROR: function: "tmedia_session_mgr_start()"
file: "src/tmedia_session.c"
line: "873"
MSG: Failed to start audio session

sip -> cisco, there's no message played, the call gets "declined" with SIP/2.0 401 Unauthorized.

Thanks.

Hi Cristi, Check this post ,

Hi Cristi,

 

Check this post , it matches the issue that you are facing.

https://code.google.com/p/webrtc2sip/wiki/Building_Source_v2_0

(Comment by ducdung0...@gmail.com, Jan 16, 2013 )

 

Rate useful post.

Thanks

Manish

New Member

where exactly? I know this

where exactly? I know this post but it's nothing related to this issue.

I've tested the same config with asterisk and it works.

Thanks.

New Member

Any thoughts?Thanks 

Any thoughts?

Thanks
 

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