Sip to sip with no sound

hi, i have an asterisknow server with a public ip on a datacenter in Canada.
Also i have 2 remote users (another country), behind a Router with NAT, these 2 use private IPs, while the router has the Public WAN; they are using zoiper(idefisk) with SIP to connect to the asterisk box, stun is setup with idefisk stun server as default, also, on the users account in asterisk I checked NAT for both; the users register (i can see it on the asterisk console) and are able to hear the voicemail and whatever message they have there, however, when i try to make a call from one user to the other, they are not able to hear each other. The call goes trough and I can answer or make the call properly…so does the other user, but we can not hear each other.

When voicemail is working but normal calling is not working my advice is to check the codecs used and available. It would help if you post the cli output of the attempt to set up a call.

i changed the codec to gsm and removed IAX from the users configuration…now it’s wonderful…
is GMS the best codec? besides g.729 of course

The best quality with the least cpu cycles used is alaw (used in Europe) and ulaw (used in usa and other parts of the world)

If bandwidth is not an issue my advice is to use alaw or ulaw (also refered to as g.711a/u). If you use a internet telephony provider most of the times the provider uses alaw or ulaw because of the cpu cycles used on the server of the provider and bandwidth is often sold together with the sip trunk.

If bandwidth is an issue g.729 is the best choice but you will need licenses. If you are satisfied with the sound quality while using the gsm codec and there are no complaints gsm is fine.

gsm works…but when i try ulaw or alaw i dont get sound or its all chopped and sounds like a going dead robot

SIP packets travel as utp packets, so if they are a lot more than our line supports, we lose some of them. So you hear choppy sound, or nothing.
GSM zips data and utp packets are less.

Better to use IAX extn instead of SIP.Hope using IAX ext u can call and hear together.Please mail to me if any help needs.
kgs_namboothiri M.C.A.,CIC,DISAM,CCNA,RHCE,CISA

[quote=“kgs_namboothiri”]Better to use IAX extn instead of SIP.Hope using IAX ext u can call and hear together.[/quote]I’ve already noted that IAX2 is working fine. The problem is with SIP hard phones. I can’t change them to IAX2, can I?