Troubleshooting SIP Signaling for Call Issues

Investigating SIP Signaling for Call Issues

Question

End users at a new site report being unable to hear the remote party when calling or being called by users at headquarters.

Calls to and from the PSTN work as expected.

To investigate the SIP signaling to troubleshoot the problem, which field can provide a hint for troubleshooting?

Answers

Explanations

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A. B. C. D.

C.

The issue at hand is related to audio not being transmitted properly between end-users at a new site and users at headquarters. To troubleshoot this issue, the SIP signaling needs to be investigated. SIP (Session Initiation Protocol) is a communication protocol used in VoIP (Voice over Internet Protocol) to initiate, modify and terminate multimedia sessions.

In SIP, SDP (Session Description Protocol) is used to describe the characteristics of the multimedia stream, such as the codec used for voice encoding, the transport protocol, and the IP addresses and port numbers for the media stream.

To investigate the SIP signaling and troubleshoot the problem of audio not being transmitted, one can examine the SDP content. The SDP content contains information about the media stream, including IP addresses, codecs, and transport protocols used for voice transmission.

In this context, options C and D are the most relevant fields in the SDP content that can provide a hint for troubleshooting. The "o=" line (Option C) contains the originator of the session, which includes the IP address and port number. The "c=" line (Option D) contains the connection data, which includes the IP address and port number for the media stream.

Examining the "o=" and "c=" lines can help identify any issues with the IP addresses or port numbers used for the media stream. For example, if the IP addresses are incorrect, the media stream may not be established correctly, resulting in no audio being transmitted. Similarly, if the port numbers are blocked or in use by other applications, the media stream may not be established correctly, resulting in no audio being transmitted.

Options A and B relate to headers in the SIP 200 OK response. The "Contact" header (Option A) contains the SIP URI of the endpoint that sent the response, while the "Allow" header (Option B) lists the SIP methods that the endpoint supports. While these headers may provide some information about the endpoint, they are unlikely to provide any clues for troubleshooting the audio transmission issue.