Some routers and firewalls have SIP and H.323 ALG capabilities. ALG is also referred to as
Fixup, Inspection, Application Awareness, Stateful Packet Inspection, Deep Packet Inspection,
and so forth. This means that the router/firewall is able to identify SIP and H.323 traffic
as it passes through and inspect, and in some cases modify, the payload of the SIP and
H.323 messages. The purpose of modifying the payload is to help the H.323 or SIP application
from which the message originated to traverse NAT;
Source:CCNP
Collaboration
Cloud and
Edge Solutions
CLCEI 300-820
Official Cert Guide
A. The far end connection data (c=) in the SDP was overwritten by deep packet inspection in the call signaling path.
Explanation:
RTP traffic may fail to reach the far endpoint if the c= line in the SDP (which specifies the media IP address) is modified or overwritten by deep packet inspection (DPI) in the signaling path. This can cause the media to be sent to an incorrect or unreachable IP address, leading to one-way audio issues. Other options, such as firewall blocking, would typically involve UDP ports, not TCP ports.
A. Is the only reasonfull answer and is also seen in the wild as a common issue, if the sender sends to a wrong destination no audio will be receieved.
B. Only if a is happening, normaly if MTP was invoked successfull RTP goes to MTP and then destination.
C. It is arriving so not matching the questions.
D. Normally this is no problem as RTP should be send via UDP, if it happens by TCP there is already something wrong, also this reference to Cisco most used UDP RTP range the answer makes sense in a weired way but not really due to TCP being mentioned.
Answer is C
"If the jitter is so large that it causes packets to be received out of the range of this buffer, the out-of-range packets are discarded and dropouts are heard in the audio. "
https://www.cisco.com/c/en/us/support/docs/voice/voice-quality/18902-jitter-packet-voice.html
Answer C:
Check Jitter section in the link given
https://www.cisco.com/c/en/us/support/docs/voice-unified-communications/unified-ip-phone-7900-series/7415-telecaster-trouble.html
I'm going with C as well. If the buffer is too small at the receiving end, it will discard the packets. Thus, the RTP traffic will not be received by the end device.
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