pom-ng:analyzer:rtp
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pom-ng:analyzer:rtp [2015/07/10 08:09] (current) gmsoft created |
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| + | ====== Analyzer rtp ====== | ||
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| + | 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. | ||
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| + | ===== Events ===== | ||
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| + | ^ Name ^ Payload associated ^ Description ^ | ||
| + | |rtp_stream|yes|Provide information about the RTP stream.| | ||
| + | |||
| + | ==== rtp_stream ==== | ||
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| + | This event starts when the stream begins and ends with it. | ||
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| + | ^ 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.| | ||
pom-ng/analyzer/rtp.txt ยท Last modified: 2015/07/10 08:09 by gmsoft
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