Check List for SIP Call Issues – Yeastar Support

Checklist for voice issue

Confirm the problem is no audio or one-way audio if is one-way audio which side can’t hear the audio.
Confirmed that NAT settings on PBX are correct, like external IP address and local network identification.
Confirmed that VoIP traffic is being routed correctly on the router, make sure the SIP messages and RTP stream can reach to PBX from outside network.
Confirmed that PBX can send and receive RTP stream by capturing pcap file on PBX.

Confirm the extension status or trunk status are registered
Confirm the sip call flow is correct, check with the figure according to the SIP call flow chapter
Confirm the SRTP settings match between IP phone and PBX
Confirm the codecs have been selected properly on both IP phone and PBX, the codecs should have the intersection between IP phone and PBX.

1) internal calls

2) remotely calls

3) sip trunk

1) Local endpoint IP

2) NAT external IP

3) IP address in INVITE sdp for Caller

4) IP address in 200OK sdp for callee

1) Internal calls SDP and contact header should use local private IP address

2) Calls with endpoints in different network SDP and contact header should use external IP address

Pcap file capture on PBX should find RTP stream send and receive from PBX

If problem is with sip outbound call register this trunk on IP Phone to test
Check Router or Firewall

Capture packets or logs on router prove that device can allow VOIP traffic reach to PBX

About how to get pcap file please refer to the link below:

About how to analyze sip calls in pcap file please refer to the link below: