300-815 · Question #19
End users in 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 signalin
The correct answer is D. c= line of SDP content. The c= (connection data) line in the SDP body of a SIP message contains the IP address to which RTP media should be sent. One-way audio between sites is a classic symptom of an incorrect or unreachable IP address in the SDP - for example, a private/NAT address being advertised ac
Question
Options
- AContact: header of the 200 OK response
- BAllow: header if the 200 OK response
- Co= line of SDP content
- Dc= line of SDP content
How the community answered
(46 responses)- A7% (3)
- B2% (1)
- C13% (6)
- D78% (36)
Explanation
The c= (connection data) line in the SDP body of a SIP message contains the IP address to which RTP media should be sent. One-way audio between sites is a classic symptom of an incorrect or unreachable IP address in the SDP - for example, a private/NAT address being advertised across a routed WAN, or the wrong interface IP being used. Checking the c= line in the 200 OK (or INVITE) lets you verify whether the IP address being advertised is correct and reachable from the remote site. The Contact header is used for signaling routing, the Allow header lists supported methods, and the o= line is the session origin identifier - none of these directly affect RTP media delivery.
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