sangsoo
Member
Hello,
We are currently building a web call service using WCS 5.3 and Asterisk 22.7.0.
It worked well within our internal corporate network, but after deploying the two servers on separate public networks,
we identified an issue during the SIP registration process.
What could be the cause of this issue, and how can it be resolved?
Version info
First, the firewall was configured so that WCS 5.3 and Asterisk can communicate with each other using both private IPs and public IPs.
How can this issue be resolved?
I would appreciate your response.
Best Regards.
We are currently building a web call service using WCS 5.3 and Asterisk 22.7.0.
It worked well within our internal corporate network, but after deploying the two servers on separate public networks,
we identified an issue during the SIP registration process.
What could be the cause of this issue, and how can it be resolved?
Version info
- wcs_version = 5.3.254-a6a39967ff4f7dc2f1dc38f00d9b16f1509e0f75
- wcs_client_version = 2.0.271-3af3e994e325b22388d3f8f04f80ae76e4c457e6
First, the firewall was configured so that WCS 5.3 and Asterisk can communicate with each other using both private IPs and public IPs.
- WCS : UDP ports 30000–33000 are open
- Asterisk : UDP port 5060, UDP ports 10000–20000 are open
- No additional NAT-related settings were configured in pjsip.conf on Asterisk
Code:[transport-udp] type=transport protocol=udp bind=0.0.0.0:5060
- When the Asterisk public IP is entered as the SIP Domain / SIP Outbound Proxy value on the Phone UI sample page,
-> SIP registration succeeds.
However, after the call is successfully established, when the callee (registered user) hangs up, no hangup event is received on the caller side.(This is also an issue.)
In contrast, when the caller hangs up, the hangup event is received normally. - When the Asterisk private IP (10.201.201.86) is entered as the SIP Domain / SIP Outbound Proxy value on the Phone UI sample page,
-> SIP registration fails.
astersik "pjsip show contacts" Unavail : (10.182.10.135 is wcs private IP)
In the WCS server logs, messages indicate that the SIP User Agent cannot be found.
Code:<-------------------- OPTIONS sip:2001@10.182.10.135:31008 SIP/2.0 from: /10.201.201.86:5060 to: /10.182.10.135:31008 time: 1768905119937 timeStamp: isSender: false transactionId: z9hg4bkpj6fcb5f22-239c-4288-aae2-944f1c65fdca callId: 29593d6e-b6b1-4e08-85ce-e4b2f445e6c1 OPTIONS sip:2001@10.182.10.135:31008 SIP/2.0 Via: SIP/2.0/UDP 10.201.201.86:5060;rport=5060;branch=z9hG4bKPj6fcb5f22-239c-4288-aae2-944f1c65fdca;received=10.201.201.86 From: <sip:2001@10.201.201.86>;tag=619be8a6-80ca-4a64-9bdf-122ea2b2e75e To: <sip:2001@10.182.10.135> Contact: <sip:2001@10.201.201.86:5060> Call-ID: 29593d6e-b6b1-4e08-85ce-e4b2f445e6c1 CSeq: 65124 OPTIONS Max-Forwards: 70 User-Agent: Asterisk PBX 22.7.0 Content-Length: 0 10:31:59,937 INFO SipUserAgentListener - EventScannerThread-34 Requested by uri sip:2001@10.182.10.135:31008 SipUserAgent null 10:31:59,937 INFO SipUserAgentListener - EventScannerThread-34 SipUserAgent not found for incoming request (Send 404 to server)- OPTIONS sip:2001@10.182.10.135:31008 SIP/2.0 Via: SIP/2.0/UDP 10.201.201.86:5060;rport=5060;branch=z9hG4bKPj6fcb5f22-239c-4288-aae2-944f1c65fdca;received=10.201.201.86 From: <sip:2001@10.201.201.86>;tag=619be8a6-80ca-4a64-9bdf-122ea2b2e75e To: <sip:2001@10.182.10.135> Contact: <sip:2001@10.201.201.86:5060> Call-ID: 29593d6e-b6b1-4e08-85ce-e4b2f445e6c1 CSeq: 65124 OPTIONS Max-Forwards: 70 User-Agent: Asterisk PBX 22.7.0 Content-Length: 0
How can this issue be resolved?
I would appreciate your response.
Best Regards.
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