| Name | Modified | Size | Downloads / Week |
|---|---|---|---|
| Parent folder | |||
| README-changelog.txt | 2011-02-22 | 2.3 kB | |
| ReleaseNotes.txt | 2011-02-22 | 2.3 kB | |
| voipmonitor_2.1-lenny.deb | 2011-02-22 | 1.5 MB | |
| voipmonitor-2.1-static.tar.gz | 2011-02-22 | 1.4 MB | |
| voipmonitor-2.1.tar.gz | 2011-02-22 | 167.6 kB | |
| Totals: 5 Items | 3.1 MB | 0 | |
New features: - WAV convertor can now handle codec changes during call, reinvite or without reinvite. It also covers cases whan each direction have different codecs. All streams are combined together and is perfectly synchronised. DTMF is also handled well. Free version supports G.711, commercial supports G.729/G.723/iLBC/GSM/Speex contact support@voipmonitor.org - voipmonitor now requires libpcap >= 1.0.0 which implements ring buffer. This buffer is storing all packets incoming on ethernet interface so voipmonitor does not miss single packet on CPU/IO peaks. If so, it is logged to syslog. Hadrcoded value is 5MB which should be enough. It can be tweak in source code. Default ring buffer in libpcap < 1.0.0 relies on /proc/sys/net/core/rmem_default which differs system to system and is low in general for accurate sniffing higher traffic (on debian etch it is 135KB). Thanks to AronHopkins (https://sourceforge.net/tracker/?func=detail&aid=3109439&group_id=312498&atid=1315315) - implement -c option for not storing CDR into databse. It is usefull whan converting pcap to WAV etc. Bug fixes: - fix MySQL crashes and garbled values. Fix useragent cdr field which was sometimes uninitialized thus empty or garbled with random data due to bad logic in code. This would also cause crashes in MySQL++. - fix open files leak introduced in version 2.0 when converting to WAV files. It caused stop writing any file to disk. - stored pcap: do not modify SDP body after rtpmap\r\n, \r was substituted to \0 which can make problems parsing by voipmonitor again and possibly another SIP parsers (wireshark is ok). If affects only if recording to wav with dynamic payload types - make sure "iscaller" is detected properly and not based on first INVITE as there can be reINVITES - do not put invalid SSRC into call table, this was causing nasty bugs if converting to wav - to minimize sync delay for wav convertor change jitter from fixed to adaptive - Whan recording calls to raw(wav) jitterbuffer was not flushing remaining packets. If there were several rtp payload changes it leads to out of sync audio. - 5 first frames were always ignored and this leads to the same issue - out of sync and missing audio - always set fbasename - fix "a_ is always caller, so check if we need to swap indexes" logic which was completely wrong.