Validating Zendesk SIP-IN from Linphone: Observing SIP and SDP with Wireshark

Introduction

When considering a telephone response system, preparing carrier lines or a PBX for testing takes time. With Zendesk Talk's SIP-IN, you can receive incoming calls from external sources using SIP. This article confirms that calls can be made from a SIP client (Linphone) on Windows and received by Zendesk SIP-IN.

What is Zendesk

Zendesk is a customer service platform for handling inquiries. In addition to ticket management and help center capabilities, Zendesk Talk allows you to handle phone support from the same interface.

What is SIP

SIP (Session Initiation Protocol) is a protocol for controlling calls. A call begins with a SIP message called INVITE. When the recipient is being called, 180 Ringing is returned, and when they answer, 200 OK is returned. After the call is established, voice is transmitted via RTP (Real-time Transport Protocol). SDP is the body included in SIP messages that determines voice conditions. Information such as which codec to use and which IP and port to receive voice on is written in the SDP.

Testing Environment

• Windows 11
• Linphone Desktop 5.2.6 (Core 5.3.72)
• Wireshark 4.6.3

Target Audience

• Those who want to quickly verify connectivity with Zendesk Talk SIP-IN
• Those who have heard SIP terminology but aren't confident in how to view it in Wireshark
• Those who want to get a basic understanding of operation before production implementation

References

• Zendesk: Adding a SIP-IN line
• Linphone Docs: Make a call
• Wireshark User Guide

Overall Process and Preparation

Create a SIP-IN line on the Zendesk side and make a call from Linphone to the SIP URI. After establishing the call, verify SIP and SDP with Wireshark.

Prerequisites

• Access to a Zendesk account with Zendesk Talk
• Administrative rights to operate the Admin Center
• A SIP account that can be used with Linphone

Zendesk Preparation

1. Create a SIP-IN line in Admin Center
   

2. Enter configuration items
   

3. Note the SIP domain (e.g., example.com) (which you'll use as ZENDESK_SIP_HOST in later steps)

Wireshark Preparation

1. Install Wireshark
2. Select your normal network connection (Wi-Fi or Ethernet) as the interface to capture
   
3. Enter sip || sdp || rtp in the display filter
   

Making a Call from Linphone to Zendesk's SIP URI

Verify that the SIP client can reach the SIP-IN entrance.

Linphone Preparation

Sign in to a SIP account in Linphone to make a call. If you already have a SIP account, use it. If not, set up a test SIP account. The account doesn't need to be related to Zendesk.

Making the Call

1. Start Wireshark capture
2. In the Linphone dialer, enter sip:ZENDESK_SIP_HOST;transport=tcp (e.g., sip:example.com;transport=tcp)
   
3. Make the call
4. Confirm that an incoming call appears on the Zendesk side
   
5. Answer and confirm that the call is established
6. End the call and stop the Wireshark capture
7. After the call, confirm that transcription is generated in Zendesk
   

Observations

The following logs were observed in Wireshark:

The following was confirmed:

• INVITE was sent
• 100 Trying was returned
• 180 Ringing was returned
• 200 OK was returned

Next, examine the SDP content returned from the Zendesk side.

Zendesk officially documents support for G.711 μ-law (PCMU) and A-law (PCMA). Looking at the actual SDP content, the 200 OK (INVITE) SDP includes a=rtpmap:0 PCMU/8000 and a=rtpmap:8 PCMA/8000. This confirms that the Zendesk side accepts G.711 μ-law (PCMU) and A-law (PCMA).

Findings from the Test

In this test environment, SIP responses were not observed when initiating via UDP. However, when specifying transport=tcp, everything from 100 Trying to 200 OK was received, and the call was established. When verifying SIP connectivity, protocol differences may impact results even when connecting to the same destination.

When the agent status was offline, the system routed to voicemail, and no incoming call was generated for the agent. Even if SIP logs appear normal, the operational status on the Zendesk side can affect the outcome, so it's necessary to check the Talk status when troubleshooting.

Conclusion

In this article, we tested Zendesk SIP-IN. We made a direct call from Linphone to a SIP URI and confirmed that the SIP-IN entrance was functioning. We also observed SIP and SDP in Wireshark and documented evidence of success. The test revealed that specifying TCP enabled successful call establishment and that Zendesk accepts G.711 (PCMU/PCMA).