Question

Users report that when an internal IP phone calls a remote site over a SIP trunk, the internal user can hear the remote party clearly, but the remote party hears nothing from the internal user. Signaling completes normally and the call stays connected. RTP statistics on the internal phone show packets being transmitted but zero packets received on the far end. What is the MOST likely cause of this one-way audio condition?

Accepted Answer

The RTP media stream from the internal phone toward the remote site is being dropped, typically due to a missing return route or firewall/ACL blocking that unidirectional media path

Answer

The two endpoints negotiated incompatible codecs in the SDP, causing the media session to fail to establish

Answer

The jitter buffer on the remote phone is undersized, causing late packets to be discarded

Answer

The DSCP marking on signaling packets was stripped, delaying SIP INVITE delivery