I've been troubleshooting a 1 way voice issue between a location with 7940s and calls to the PSTN. The users say they cannot hear the person they are calling. I've had local support do some wireshark captures of these calls. At first, I wasn't seeing any problem as when I do a stream analysis of the RTP traffic I hear both sides of the conversation. However, looking closer at the RTP analysis I see that Max Delta, Max Jitter, and Max Skew are extremely high with potentially almost 97% packet loss. See below for a screenshot. For reference, .11 is the CUCM Subscriber and .152 is the phone.
So my questions. If this is showing 97% lost, how can I play the stream and hear both sides with using Jitter Buffer or Uninterrupted Mode under Playback Timing. If I choose RTP Timestamp, I hear the phone side of the conversation and the ringing, but once the far end picks up (a voicemail box in this case) the conversation stops. Needless to say I'm thoroughly confused about what is happening here. I've also attached just the RTP packets of the Wireshark capture if that will help.
