This shows you the differences between two versions of the page.
— |
pom-ng:analyzer:rtp [2015/07/10 08:09] (current) gmsoft created |
||
---|---|---|---|
Line 1: | Line 1: | ||
+ | ====== Analyzer rtp ====== | ||
+ | |||
+ | This analyzer will process RTP connections and create a payload out of them. It will use the telephony API to fetch the codec information and generate the appropriate event and payload. | ||
+ | |||
+ | ===== Events ===== | ||
+ | |||
+ | ^ Name ^ Payload associated ^ Description ^ | ||
+ | |rtp_stream|yes|Provide information about the RTP stream.| | ||
+ | |||
+ | ==== rtp_stream ==== | ||
+ | |||
+ | This event starts when the stream begins and ends with it. | ||
+ | |||
+ | ^ Field ^ Type ^ Description ^ | ||
+ | |src_addr|ipv4 or ipv6|Source IPv4 or IPv6 address.| | ||
+ | |dst_addr|ipv4 or ipv6|Destination IPv4 or IPv6 address.| | ||
+ | |src_port|uint16|Source port.| | ||
+ | |dst_port|uint16|Destination port.| | ||
+ | |sess_proto|string|Session protocol (i.e. sip/mgcp/h323).| | ||
+ | |call_id|string|Call-ID to which this stream belongs.| | ||
+ | |ssrc|uint32|Synchronization source identifier.| | ||