I’ve just managed to make my first external call to one of the SIP phones on my network… this was a great breakthrough! However, the call quality on the external callers phone is really bad, with scratchy sound, clipped and choppy voices etc. The sound quality on the internal SIP phone is excellent however.
Can anybody point to some obvious ways of increasing the quality of the external user’s phone call? is there a conf file to alter the bit rate/codec etc?
All of the SIP phones are connected to the Asterisk server via a separate layer 2 gigabit switch (no other workstations are on here) and all SIP to SIP calls have no qulaity problems. the incoming call is being made through a 4 channel ISDN setup. The server is running a basic Fedora Core 4 setup with no X installed.
Thanks in advance.
John
codecs are set in sip.conf, either globally or per-device. the syntax is allow=codec-name…so you’d have
disallow=all
allow=gsm
allow=ulaw
that means that gsm is the first priority codec, followed by ulaw.
i don’t know ISDN, so i can’t comment on how that might be affecting the sound quality, but it sounds like there might be a timing issue as well.
good luck, sorry i couldn’t be more help.
[quote=“whoiswes”]codecs are set in sip.conf, either globally or per-device. the syntax is allow=codec-name…so you’d have
disallow=all
allow=gsm
allow=ulaw
that means that gsm is the first priority codec, followed by ulaw.
i don’t know ISDN, so i can’t comment on how that might be affecting the sound quality, but it sounds like there might be a timing issue as well.
good luck, sorry i couldn’t be more help.[/quote]
Thanks for that… it’s pointed me in the right direction! I’ll try and get a bit further!
Hi
It does sound like a transcoding issue, If you are dialing in and out over the isdn make sure the default is g711u for us or g711a for europe and see what its like.
Ian
[quote=“ianplain”]Hi
It does sound like a transcoding issue, If you are dialing in and out over the isdn make sure the default is g711u for us or g711a for europe and see what its like.
Ian[/quote]
Thanks Ian
So, as I’m in Europe, I would use:
allow=g711a
is this the same as: allow=alaw?
John
Yep
European ISDN should use alaw
[quote=“ianplain”]Yep
European ISDN should use alaw[/quote]
Ian
I’ve changed the allow=alaw (it was ulaw) and this seems to have improved the quality on the SIP phone, but nothing has really happened on the callers side (i.e. the external phone call) I’ve tried it using a couple of moblies through different providers and from a BT landline and the speech is very distorted… there is slightly less clipping, but huge distortion.
I’m probably way off line here but to me, altering a codec in sip.conf would surely only improve/degrade the performance of the SIP side… is there another section that would alter the encoding of the data going across the ISDN? Or does the sip.conf codec encode the SIP call before going externally?
SIP to SIP quality is excellent, when an external phone calls an internal SIP phone, the quality of the SIP phone is fine but the external phone is scratchy and distorted. Unfortunately, I haven’t managed to make an internal to external phone call yet, so I can’t comment on what the quality on that side is.
One last thing, I’m using BRIStuff 0.2.0-RC8q with Asterisk 1.0.10, will it make any difference if I upgrade to the latest BRIStuff 0.3.0-PRE-1l with Asterisk 1.2.4?
John
Just for anyone who found this topic on a search and wanted to know the outcome, all I did was down the server, re-cable the Junghanns card to the ISDN NTE and brought it back up again. Everything was sorted. The sound was crisp and crystal clear.