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View Full Version : VOIP SIP best codec for outgoing voice



Fishface
3rd August 2007, 08:56
Hello,

does anyone know the best codec to use for SIP/VOIP for outgoing voice?

I can hear people perfectly - the other side complains that they hear me quietly.

And the microphone is working and placed correctly.

Any tips on how to improve the outgoing voice quality?

cojak
5th August 2007, 13:51
Get a proper phone.

I'm using the Linksys SPA941 - much better sound quality than a microphone.

I use a headset from it if I need hands free.

NoddY
5th August 2007, 20:18
g711 aLaw

see http://www.voip-info.org/wiki/view/ITU+G.711

Fishface
6th August 2007, 07:35
Ta Noddy - done the change - no more complaints yet.

I am using a Draytek Vigor 2800vg - excellent - got the telephone plug ot a decent normal phone using a SIP service.

Those IP phones are gy-nor-mouse - they have a look of Sugar's Daftphone

Any idea what codec Skype uses? - I have noticed a deterioration of their service for some time now - cant find their advanced settings.

NoddY
6th August 2007, 09:35
...Any idea what codec Skype uses? - I have noticed a deterioration of their service for some time now - cant find their advanced settings.

Skype is very complex. Nobody knows what codec(s) Skype are using because it's proprietary and I believe it dynamically adjusts its codecs per call!

Skype is a peer-to-peer system and will use a series of clever tricks to circumvent firewalls and to deal with NAT, such as tunnelling via HTTP and STUN. A 'local' group of Skype users establish a mesh, a few of which become 'supernodes'. All this carry-on is transparent to the end user. In other words, it's very difficult to pinpoint bottlenecks.

I recommend sticking to SIP/RTP [or IAX2 which is firewall and bandwidth friendly].

Churchill
6th August 2007, 10:08
Skype is very complex. Nobody knows what codec(s) Skype are using because it's proprietary and I believe it dynamically adjusts its codecs per call!

Skype is a peer-to-peer system and will use a series of clever tricks to circumvent firewalls and to deal with NAT, such as tunnelling via HTTP and STUN. A 'local' group of Skype users establish a mesh, a few of which become 'supernodes'. All this carry-on is transparent to the end user. In other words, it's very difficult to pinpoint bottlenecks.

I recommend sticking to SIP/RTP [or IAX2 which is firewall and bandwidth friendly].

That's what RTCP is for...

Fishface
6th August 2007, 11:25
[QUOTE=NoddY] A 'local' group of Skype users establish a mesh, a few of which become 'supernodes'.

QUOTE]

You mean each Skype user is carrying other peoples calls through the skype on their computer? thus the deterioration of quality? buzz on the line etc

Hmmmn wonder how many...

NoddY
6th August 2007, 12:20
[QUOTE=NoddY] A 'local' group of Skype users establish a mesh, a few of which become 'supernodes'.

QUOTE]

You mean each Skype user is carrying other peoples calls through the skype on their computer? thus the deterioration of quality? buzz on the line etc

Hmmmn wonder how many...


It's likely that low latency nodes, especially un-NATted ones, without restrictions on UDP traffic and significant uptime, would be elevated in the Skype hierarchy. Therefore such nodes are likely to carry multiple conversations, or assist in the setting up of other people's calls. Encryption prevents eavesdropping.

bored
6th August 2007, 14:24
Skype is a peer-to-peer system and will use a series of clever tricks to circumvent firewalls and to deal with NAT, such as tunnelling via HTTP and STUN. A 'local' group of Skype users establish a mesh, a few of which become 'supernodes'. All this carry-on is transparent to the end user. In other words, it's very difficult to pinpoint bottlenecks.


One of their "clever tricks" is to quietly take over port 80 without asking the user :tantrum: Took me quite a while to figure out why the app server on a demo machine would sometimes stop working after a reboot ...