Hi! My personal experience says G711, G729 and GSM with 20 ms of payload per packet are the most used ones. This leads to IP packets (incl. IP header) with the following size (xlite+sipura): G711 20ms 200 Byte G729 20ms 60 Byte GSM 20ms 73 Byte Microsoft Messenger uses for Messenger<->Messenger conversations the SIREN Codec with 16kHz = very good quality :-) SIREN 20ms 80 Byte Other Codecs like ilbc and speex are sometimes supported, but I never saw an application using them (without explicit forcing them) Nevertheless, here are the details (tested with xlite): iLBC 240 20/40ms 90 Bytes speex 160 20ms 86 Bytes xlite sends ilbc packets alternating every 20 and 40 ms - I don't know if this is because of the codec or the implementation within xlite. So, I suggest these Codecs as "must". Maybe bigger packets with a payload of 40ms might be interesting. The number of packets depends on the simulated call duration - I'm sure there will be some papers stating: 95% of calls are XX minutes long. I suggest to simulate short calls (30, or 60 seconds) and maybe some long calls, e.g. 15 minutes (to see more packet loss bursts). regards Klaus Furthermore
-----Original Message----- From: Daniel Karrenberg [mailto:dfk@ripe.net] Sent: Tuesday, May 25, 2004 2:16 PM To: Alexander Mayrhofer Cc: Klaus Darilion; NETZER Kurt; Michael Haberler; Henk Uijterwaal Subject: Re: VoIP Messungen
Meine Erfahrung ist, dass bei VAD das "Einschalten" oft Probleme macht. Ob das daran liegt, dass der Stream beim Wiederanlaufen behindert wird oder am Codec weiss ich nicht. Ich gucke mir das Paper noch an. Was wir brauchen sind Vorschlaege, welche Parameter wir fuer unsere Streams benutzen sollen:
- Packetgroaesse - Intervalle - Anzahl Packete
Daniel
PS: Die mailinglist ist fertig. To susbscribe measure-voip@ripe.net. Aber bitte in Englisch posten, da Andere vielleicht auch mitmachen wollen.
Daneil