Check List for SIP Call Issues

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.

Checklist for call establishment issue

  • 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.

Troubleshooting approach

  1. Scenario

1) internal calls

2) remotely calls

3) sip trunk

  1. List IP Address

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. Check IP Address in SDP and contact header

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

  1. Check RTP Stream on PBX

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

  1. If problem is with sip outbound call register this trunk on IP Phone to test
  2. 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:

https://support.yeastar.com/hc/en-us/articles/360007739473-How-to-Get-PCAP-Dump-Capture

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

https://support.yeastar.com/hc/en-us/articles/360007606533-How-to-Analyze-SIP-Calls-in-Wireshark

cases:

1. One-way audio with Cisco CUCS

Check the call flow in pcap file, SDP offer of cisco is in the ACK request instead of INVITE request

Root cause: PBX now not support SDP in ACK

Solution:

  1. enable Media Termination Point Required if you have MTP on CUCS
  2. enable Early Offer Support for Voice and Video Calls if you don’t have MTP

2. No voice after Linkus establish call with IP Phone in same local network

Check translation path of ilbc codec in Asterisk CLI

IPPBX*CLI> core show translation paths ilbc
--- Translation paths SRC Codec "ilbc" sample rate 8000 --- 
ilbc:8000 To g723:8000 : No Translation Path 
ilbc:8000 To ulaw:8000 : No Translation Path 
ilbc:8000 To alaw:8000 : No Translation Path 

Root cause: DSP issue

Solution: submit a ticket to yeastar

3. One way audio/call get disconnected for inbound call through sip trunk

Check the call flow in pcap file, there is re-invite request send from PBX right after call established.

Root cause: provider do not support re-invite request

Solution: disable Allow re-invite in Settings > PBX > General > SIP > Advanced

4. SIP trunk outbound call failed

Check the INVITE sip message send out from PBX, if there is Diversion header in there.

Root cause: Diversion header should be used when the call is be forwarded.

Solution: disable Diversion in sip trunk

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