Chapter 9. Telephony

Table of Contents

9.1. Introduction
9.2. Playing VoIP Calls
9.2.1. Supported codecs
9.2.2. Work with RTP streams - Playlist
9.2.3. Playing audio during live capture
9.2.4. RTP Decoding Settings
9.2.5. VoIP Processing Performance and Related Limits
9.3. VoIP Calls Window
9.4. ANSI
9.4.1. A-I/F BSMAP Statistics Window
9.4.2. A-I/F DTAP Statistics Window
9.5. GSM Windows
9.6. IAX2 Stream Analysis Window
9.7. ISUP Messages Window
9.8. LTE
9.8.1. LTE MAC Traffic Statistics Window
9.8.2. LTE RLC Graph Window
9.8.3. LTE RLC Traffic Statistics Window
9.9. MTP3 Windows
9.10. Osmux Windows
9.11. RTP
9.11.1. RTP Streams Window
9.11.2. RTP Stream Analysis Window
9.11.3. RTP Player Window
9.12. RTSP Window
9.13. SCTP Windows
9.14. SMPP Operations Window
9.15. UCP Messages Window
9.16. H.225 Window
9.17. SIP Flows Window
9.18. SIP Statistics Window
9.19. WAP-WSP Packet Counter Window

9.1. Introduction

Wireshark provides a wide range of telephony related network statistics which can be accessed via the Telephony menu.

These statistics range from specific signaling protocols, to analysis of signaling and media flows. If encoded in a compatible encoding the media flow can even be played.

The protocol specific statistics windows display detailed information of specific protocols and might be described in a later version of this document.

Some of these statistics are described at the https://gitlab.com/wireshark/wireshark/wikis/Statistics pages.