Connecting SIP Voxbone on IP Office

[telepaul]

Hi,

Someone hase experience connecting inbound SIP DDI from voxbone to a SIP trunk on a IP 500V2 V6.0

know that this is only inbound traffic and that there is no authentication server at voxbone.

I have +/-30 seconds of one-way outbound voice traffic. It is not a firewall problem all other SIP traffic is working.

I think that IPO is not accepting unauthenticated traffic on the sip trunk. SIP/2.0 403 Forbidden

Any sugestions?

Thanks

Paul


Monitor trace of incoming voxbane call.

4865999mS SIP Call Rx: 18
INVITE sip:323808XXXX@81.246.8.13 SIP/2.0
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
Max-Forwards: 69
Content-Type: application/sdp
Contact: <sip:0@81.201.82.45:5060;transport=udp>
User-Agent: Vox Callcontrol
Content-Length: 309

v=0
o=root 503 503 IN IP4 81.201.82.15
s=session
c=IN IP4 81.201.82.15
t=0 0
m=audio 18456 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSuppff - - - -
a=ptime:20
a=sendrecv
4866003mS SIP Call Tx: 18
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

4867057mS SIP Call Tx: 18
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

4872331mS SIP Call Tx: 18
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3378312775 2705272364 IN IP4 192.168.1.61
s=Session SDP
c=IN IP4 192.168.1.61
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4874332mS SIP Call Tx: 18
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3378312775 2705272364 IN IP4 192.168.1.61
s=Session SDP
c=IN IP4 192.168.1.61
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4878332mS SIP Call Tx: 18
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3378312775 2705272364 IN IP4 192.168.1.61
s=Session SDP
c=IN IP4 192.168.1.61
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4886332mS SIP Call Tx: 18
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3378312775 2705272364 IN IP4 192.168.1.61
s=Session SDP
c=IN IP4 192.168.1.61
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4891584mS SIP Reg/Opt Tx: 18
OPTIONS sip:Unknown@voxbone.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;rport;branch=z9hG4bKb3f1968b89a96bc234ea0877d935054b
From: <sip:Unknown@voxbone.com>;tag=c4f816cbde8f3f1d
To: <sip:Unknown@voxbone.com>
Call-ID: 5f1010847bb4fdf09974c066109c031a@192.168.1.61
CSeq: 1294127097 OPTIONS
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

4891611mS SIP Reg/Opt Rx: 18
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.61:5060;received=81.246.8.13;branch=z9hG4bKb3f1968b89a96bc234ea0877d935054b;rport=5060
From: <sip:Unknown@voxbone.com>;tag=c4f816cbde8f3f1d
To: <sip:Unknown@voxbone.com>
Call-ID: 5f1010847bb4fdf09974c066109c031a@192.168.1.61
CSeq: 1294127097 OPTIONS
Content-Length: 0

4902332mS SIP Call Tx: 18
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 81.201.82.45:5060;branch=z9hG4bKfc6a9440c1137b5edde9ecc9926a755a
From: "anonymous" <sip:0@voxbone.com>;tag=23323
To: <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 102 INVITE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Type: application/sdp
Content-Length: 227

v=0
o=UserA 3378312775 2705272364 IN IP4 192.168.1.61
s=Session SDP
c=IN IP4 192.168.1.61
t=0 0
m=audio 49154 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
4902432mS SIP Call Tx: 18
BYE sip:0@81.201.82.45:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.1.61:5060;rport;branch=z9hG4bK14b73647114c458fb23ed8b35ba4e50c
From: "323808XXXX" <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
To: "anonymous" <sip:0@voxbone.com>;tag=23323
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 103 BYE
Contact: "323808XXXX" <sip:323808XXXX@192.168.1.61:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, INFO, UPDATE
Supported: timer
Content-Length: 0

4902467mS SIP Call Rx: 18
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.61:5060;received=81.246.8.136;branch=z9hG4bK14b73647114c458fb23ed8b35ba4e50c;rport=5060
From: "323808XXXX" <sip:323808XXXX@81.246.8.13>;tag=ed6da124d1be3e78
To: "anonymous" <sip:0@voxbone.com>;tag=23323
Call-ID: 2G7XKI5R7VGNVKANVO5GWCHMXQ@81.201.82.45
CSeq: 103 BYE
Content-Length: 0

 

[IPOfficeIreland]

seems to be an issue with v6. did this trunk work on v5?

The invite is received, the IPO Trys to connect and in the TX Ringing 180 message sends back the Contact header with the IP to use for RTP

SIP/2.0 180 Ringing Via: SIP/2.0/UDP Contact: "323808XXXX">sip:323808XXXX@192.168.1.61:5060;transport=udp>

You note the 192.168.1.61.
As HSM says this can be down to the router transforations (

http://www.tek-tips.com/viewthread.cfm?qid=1596914&page=2)

so this is worth checking.

How is your network set up?

What do you have in the SIP line settings and the LAN port Firewall settings, STUN, etc?

 

[IPOfficeIreland]

enable all SIP traces in Monitor (except the 2 x HEX), and enable all in SYSTEM. Post the full output, this will show STUN etc.

 

[telepaul]

I have 3 other sip providers working inbound and outbound. without any problems. Only this one with voxbone gives me trouble. This provider just routed incoming calls to a public IP. You need to created your URI on there website, to link the international DDI's, as they explain on the next question.

What do I need to use the Voxbone service ?
=> You need a SIP compatible equipment with a public IP address and you need to be capable of receiving calls without being registered. Voxbone does not provide SIP registration nor NAT traversal service.

As the 3 other providers are working well, it looks that it is not a NAT problem.

It looks that voxbone has a problem with the "contact" in the header that points to a local IP, as the "to" and "from" are pointing to a public IP.

As they said, Voxbone does not provide NAT traversal.

 

[TheSmash]

One way speech is usually a routing issue.

I know you say you have other providers working but that doesn't eliminate Router/firewall.

I have a suggestion for you as I noticed strange things when configuring multiple providers. Under your SIP trunk, set Use Network Topology to None on all your SIP trunks.

See what difference that makes...

My name is Mike but everyone calls me The Smash...

 

[telepaul]

All SIP trunks are already set to none for the "Use Network Topology".

I will "wireshark" the LAN en WAN to see what is missing.