SIP trunk inbound trace

[Alfalis]

Hey guys,

I ran into a strange problem while installing an IPO with a SIP trunk:
Outbound calls work fine, but inbound don't.

What I get for an inbound call looks like this:
SIP Rx
from callingnumber@sip_provider_ip:5061
to callednumber@public_ip:1279
invite

SIP Tx
from callednumber@sip_provider_ip:5061
to callednumber@public_ip:1279
trying

IPO is then looking up the call route and even ringing the desired internal extension (and sending ringing in the same way as the trying) but when you pick it up you can't hear anything. In the meantime the provider keeps sending invites as if he never got the response trying and ringing from the IPO.

My question now is if the responses my IPO is sending are generally correct (from and to fields) and I might have to look into the network/firewall or if my sip trunk configuration is wrong.

Thanks in advance for your help!

 

[janni78]

You're not showing the complete trace or SIP packet so at best I can guess.

"Trying is the first step to failure..." - Homer

 

[rdoubrava]

How is the SIP trunk setup?
What are you settings in SIP URI?

 

[Alfalis]

@Janni
Here you go, initial INVITE and the IPOs TRYING (IPs and numbers replaced with square brackets), after this the provider continues sending identical invites:

10:51:35 518499mS SIP Rx: UDP [provider_IP]:5061 -> [IPO LAN2 IP]:5060

INVITE sip:[called_number/SIP contact]@[customer_external_IP]:1279;user=phone SIP/2.0​

Via: SIP/2.0/UDP [provider_IP]:5061;rport;branch=z9hG4bK-1758154314-3859914247-2823073936-999632194​

From: <sip:[calling_number]@[provider_IP]:5061;user=phone>;tag=4240002634-3859914247-2823073936-999632194​

To: <sip:[called_number/SIP contact]@[customer_external_IP]:1279;user=phone>​

Call-ID: 4a4eba1a079e11e690b044a8422d953b@[provider_IP]​

CSeq: 1 INVITE​

Contact: <sip:[calling_number]@[provider_IP]:5061;user=phone>​

Content-Type: application/sdp​

Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, UPDATE​

Max-Forwards: 70​

User-Agent: TS-v4.5.1-20g​

Cisco-Guid: 2730798765-628772519-609419767-609419767​

Remote-Party-ID: <sip:[calling_number]@[provider_IP]:5061;user=phone>;party=calling;privacy=off;screen=yes​

Content-Length: 255​

v=0​

o=- 1461228711 1461228711 IN IP4 94.140.90.241​

s=-​

c=IN IP4 94.140.90.241​

t=0 0​

m=audio 29194 RTP/AVP 8 0 101​

a=rtpmap:8 PCMA/8000​

a=rtpmap:0 PCMU/8000​

a=rtpmap:101 telephone-event/8000​

a=fmtp:101 0-15​

a=sendrecv​

a=silenceSuppff - - - -​

10:51:35 518503mS CMCallEvt: 0000000000000000 0.1012.0 -1 BaseEP: NEW CMEndpoint f4fb8b44 TOTAL NOW=1 CALL_LIST=0
10:51:35 518506mS SIP Tx: UDP [IPO LAN2 IP]:5060 -> [provider_IP]:5060

SIP/2.0 100 Trying​

Via: SIP/2.0/UDP [provider_IP]:5061;rport;branch=z9hG4bK-1758154314-3859914247-2823073936-999632194​

From: <sip:[calling_number]@[provider_IP]:5061;user=phone>;tag=4240002634-3859914247-2823073936-999632194​

Call-ID: 4a4eba1a079e11e690b044a8422d953b@[provider_IP]​

CSeq: 1 INVITE​

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE​

Supported: timer,100rel​

Server: IP Office 9.1.5.0 build 145​

To: <sip:[called_number/SIP contact]@[customer_external_IP]:1279;user=phone>;tag=d02ae1ad7e43e6ee​

Content-Length: 0​

@rdoubrava
What settings exactly do you need?
SIP URI:
Via

IPO LAN2 IP​

Local URI

Use Credentials Contact​

Contact

Use Credentials Contact​

Display Name

Use Credentials Contact​

PAI

None​

Incoming Group

0​

Outgoing Group

0​

 

[janni78]

You need to look at the SIP TX "OK" message that is sent from the IP Office when you answer the call.

"Trying is the first step to failure..." - Homer

 

[Alfalis]

@janni78
The OK Tx after picking up the call looks like this:
09:00:42 80272701mS SIP Tx: UDP [IPO LAN 2 IP]:5060 -> [provider_IP]:5060

SIP/2.0 200 OK​

Via: SIP/2.0/UDP [provider_IP]:5061;rport;branch=z9hG4bK-3903566579-3859896072-2823041664-2828152130​

From: <sip:[calling_number]@[provider_IP]:5061;user=phone>;tag=3600462579-3859896072-2823041664-2828152130​

Call-ID: f3b29aea085711e6803244a8422d92a8@[provider_IP]​

CSeq: 1 INVITE​

Contact: <sip:[called_number/SIP contact]@[IPO LAN 2 IP]:5060;transport=udp>​

Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE​

Supported: timer,100rel​

Server: IP Office 9.1.5.0 build 145​

To: <sip:[called_number/SIP contact]@[IPO LAN 2 IP];user=phone>;tag=02eb5bbc06f6695b​

Content-Type: application/sdp​

Content-Length: 208​

v=0​

o=UserA 3695567223 2350031407 IN IP4 [IPO LAN 2 IP]​

s=Session SDP​

c=IN IP4 [IPO LAN 2 IP]​

t=0 0​

m=audio 46752 RTP/AVP 8 101​

a=rtpmap:8 PCMA/8000​

a=rtpmap:101 telephone-event/8000​

a=fmtp:101 0-15​

Basically the trace looks like this:
Rx - Invite
Tx - Trying
IPO is looking up the call route
Tx - Ringing (at this point the extension is actually ringing)
Rx - Invite
Tx - Ringing
Rx - Invite
Tx - Ringing
Rx - Invite
Tx - Ringing
Now I'm picking up the phone
Tx - Ok
Rx - Invite
Tx - Ok
Caller hangs up
Rx - Cancel
Tx - Ok
Rx - Cancel
Tx - Ok
Rx - Cancel
Tx - Ok
At this point I stopped the trace

It looks a lot like the provider never receives the responses from IPO which brings me back to my original question if the trying/ringing/ok from IPO are ok or if maybe the to and from fields have wrong values.

Could anyone post an example trace from a working inbound call?

 

[janni78]

This is your issue

v=0
o=UserA 3695567223 2350031407 IN IP4 [IPO LAN 2 IP]
s=Session SDP
c=IN IP4 [IPO LAN 2 IP]
t=0 0
m=audio 46752 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

You're telling the service provider send voice to your internal IP, you need to use Network Topology so the information gets replaced with the external IP.

"Trying is the first step to failure..." - Homer

 

[janni78]

Same issue in the Contact field which probably is why your service provider keeps sending invites.

"Trying is the first step to failure..." - Homer

 

[amriddle01]

That's not always the case janni78, our SIPs allow internal addresses to be sent, the SBC (trunk side) takes care of the translation

 

[redphone]

They keep sending INVITE messages because they never see the 100 Trying. From the trace, it looks like they are listening on port 5061, but you are sending to port 5060.

10:51:35 518499mS SIP Rx: UDP [provider_IP]:5061 -> [IPO LAN2 IP]:5060

10:51:35 518506mS SIP Tx: UDP [IPO LAN2 IP]:5060 -> [provider_IP]:5060

 

[Alfalis]

@redphone
That would make sense..
I played around with the ports a little but the problem is that the trunk only registers correctly when the registration is sent from my 5060 to the providers 5060.
Once the trunk is registered all further inbound traffic from the provider is coming from 5061 to 5060 and the IPO (as you mentioned) is answering 5060 to 5060..
Any way to send only the registration to 5060 and all further traffic to 5061?

EDIT: Outbound calls still work fine, the IPOs invites go from 5060 to 5060 on the provider side and the provider is answering 5060 to 5060, the whole connection/SIP "conversation" between the two is working great outbound^^
Only problem I see now is that inbound calls are coming from 5061 :/

 

[amriddle01]

Tell the provider to sort their act out then

 

[Alfalis]

A classic amriddle01 answer, didn't expect anything less from you xD
I'll ask them, let's see what they'll respond

 

[Westi]

aren't we glad we don't have to deal with amriddle01 as a line provider

Joe W.

FHandw, ACSS (SME)


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