Voip is the best option for the tool. standard is the android application working principal.

To simulate the voice signal over different channel, the best tool (as I know) is MATLAB. For that you have to consider the channel as a time varying filter and the filter coefficients have to be set based on the type of channel you considered.   Yes, ITU-T is best or you can also consider ETSI.

Thank you very much Arun.

Do you happen to know if there are MATLAB toolboxes that can help me acomplish these simulations?

Also, do you know which ITU-T and ETSI "standards" would help me to define the filters for the channels?

For TDM you need to simulate the encoding and decoding (G.711 or G.726 for international links). You can learn about acceptable BER from G.821 and G.826. Next you need to model the effects of line echo - see G.168 and the text of the relevant P series documents.

For packet based (e.g., VoIP) systems you need to do is to implement the desired compression and decompression (e.g., G.729, G.723, AMR), and then simulate the effects of packet loss on the compressed voice packets. You need to decide whether compressed frames are sent separately or several per packet, and then to model the packet loss (the accepted way is to use a GIlbert-Eliott state machine, as detailed in ETSI and ITU-T documents).

If you need to model delay and PDV as well, you could use real traces, or generate traces using a tool like NS2 or NS3.

If you need to simulate acoustic echo, then further work is involved. Read the relevant P series documents.

Y(J)S

Thank you very much Mr. Stein.

I've been doing some researching on this and I want to summarize some of my results in case anyone faces this challenge in the future.

Plain Old Telephone Service:

I find the model proposed in this article is quite comprehensive: Dimolitsas, S.; Gunn, J.E., "Modular, offline, full duplex telephone channel simulator for high speed data transceiver evaluation," Radar and Signal Processing, IEE Proceedings F , vol.135, no.2, pp.155,160, Apr 1988

On the other hand, ITU-T Recommendation G.120 covers the basics of the Public Switched Telephone Networks. Other ITU-T Recommendations already mentioned by Mr. Stein are G.711, G.726 (codecs), G.821, G.826 (BER).

I also found several phone line emulators with line impairment features that can simulate the signal degradation of a long distance phone line, if you need a link to the websites of the makers of this equipment just send me a message.

Mobile telephone service:

3GPP standards are the main source of information here.

The main codecs used nowadays in GSM networks are

EFR: 3GPP TS 06.60 Enhanced full rate speech transcoding

AMR: 3GPP TS 26.071 Mandatory speech CODEC speech processing functions; AMR speech Codec; General description

AMR-WB: This can be found as ITU-T Recommendation G.722.2 or as 3GPP TS 26.171 Speech codec speech processing functions; Adaptive Multi-Rate - Wideband (AMR-WB) speech codec; General description

For the LTE channel model, this 3GPP document has been cited in several articles about simulations: 3GPP TR 36. 814, E-UTRA; Further advancements for E-UTRA physical layer aspects.

VoIP:

One of the main standards for this kind of communication channel is ITU-T Recommendation H.323 which also lists, among others, the codecs described in the following recommendations:G.711 (mandatory) , G.723.1, G.726,

The widely used Skype uses Silk codec, which is described in the IETF draft: SILK Speech Codec draft-vos-silk-01

Other noteworthy codecs include: Opus (described in IETF RFC 6716) and iLBC (described in ITF RFC 3951).

Some protocols worth looking for are: UDP and RTP (IETF RFC 3550).

I also found network emulators which can simulate the packet loss, jitter and other effects which disrupt VoIP communication. If you need the websites of the makers of this hardware write me a message.

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