asterisk users - Nov 2006 - SPA-941 (and others ) Transmit Sound Quality

Hello,

This is not exactly an Asterisk question, but I was encouraged to seek
advice here anyway. The kindness of the * open source community is
legendary :)

I am getting going with an Asterisk 1.2 box, and I'm having trouble
getting good quality transmit sound using handsets with VoIP phones. I'm
primarily trying to focus on SPA-941, but also experimenting with Aastra
9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the
box to provide hardware timing.

The use case is calling from the SIP phones (which are extensions
registered with the * 1.2 box) to a VoIP termination service which routes
the call to a PSTN number. Everything sounds great on the SIP phone, but
the sound on the other end of the line is distant and missing bass, most
especially so on the SPA-941 (which is the phone we really want to use).
If I use the default handset mic gain value of 0db, the sound is so loud
for the other person they have to hold the phone away from their ear. If I
set it to -6db, it is still too quiet. The Aastra 9113i sounds a little
better, and the Uniden 5.4 GHz Cordless sounds actually very good, so I'm
pretty sure my network setup is capable of transmitting good sound. Using
the speaker-phone on the SPA-941 sounds significantly better than using
the handset. But we need the handset to also sound good.

I've tried different providers etc. and always come back to the phone.
I'm
using G711u codec in all cases and silence suppresion is off.

I saw a previous thread that mentioned changing the RTP from .03 to .02,
however the post was regarding a MeetMe issue. I tried anyway, and it
introduced an echo on the line.

I've seen many rave reviews regarding the sound quality on the SPA-941, so
I'm wondering if maybe I got a bum handset? Would anyone be willing to
receive/place a call to tell me if it sounds the way its supposed to or if
there is indeed a problem?

All suggestions/recommendations greatly appreciated.

Much thanks,

-- Ron
ron@iamron.net

ron,

i recently fixed the poor quality with our spa942 using canreinvite=yes
have you found out the problem with your spa941? i can't get around the 
problem of poor quality audio when the rtp pass through *

ron@iamron.net wrote:
> Hello,
>
> This is not exactly an Asterisk question, but I was encouraged to seek
> advice here anyway. The kindness of the * open source community is
> legendary :)
>
> I am getting going with an Asterisk 1.2 box, and I'm having trouble
> getting good quality transmit sound using handsets with VoIP phones.
I'm
> primarily trying to focus on SPA-941, but also experimenting with Aastra
> 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the
> box to provide hardware timing.
>
> The use case is calling from the SIP phones (which are extensions
> registered with the * 1.2 box) to a VoIP termination service which routes
> the call to a PSTN number. Everything sounds great on the SIP phone, but
> the sound on the other end of the line is distant and missing bass, most
> especially so on the SPA-941 (which is the phone we really want to use).
> If I use the default handset mic gain value of 0db, the sound is so loud
> for the other person they have to hold the phone away from their ear. If I
> set it to -6db, it is still too quiet. The Aastra 9113i sounds a little
> better, and the Uniden 5.4 GHz Cordless sounds actually very good, so
I'm
> pretty sure my network setup is capable of transmitting good sound. Using
> the speaker-phone on the SPA-941 sounds significantly better than using
> the handset. But we need the handset to also sound good.
>
> I've tried different providers etc. and always come back to the phone.
I'm
> using G711u codec in all cases and silence suppresion is off.
>
> I saw a previous thread that mentioned changing the RTP from .03 to .02,
> however the post was regarding a MeetMe issue. I tried anyway, and it
> introduced an echo on the line.
>
> I've seen many rave reviews regarding the sound quality on the SPA-941,
so
> I'm wondering if maybe I got a bum handset? Would anyone be willing to
> receive/place a call to tell me if it sounds the way its supposed to or if
> there is indeed a problem?
>
> All suggestions/recommendations greatly appreciated.
>
> Much thanks,
>
> -- Ron
> ron@iamron.net
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

I too seem to have the same problem, dont know about "poor quality"
but its certainly not loud enough, I have to put my mouth to the
microphone, otherwise the other end reports they cannot hear me. This
does however seem to do a good job to cancel out the background noise

On 11/10/06, Ron Winograd <ron@iamron.net> wrote:
> Hello,
>
> This is not exactly an Asterisk question, but I was encouraged to seek
> advice here anyway. The kindness of the * open source community is
> legendary :)
>
> I am getting going with an Asterisk 1.2 box, and I'm having trouble
> getting good quality transmit sound using handsets with VoIP phones.
I'm
> primarily trying to focus on SPA-941, but also experimenting with Aastra
> 9113i and Uniden UIP1868. I do not at this time have any PSTN cards in the
> box to provide hardware timing.
>
> The use case is calling from the SIP phones (which are extensions
> registered with the * 1.2 box) to a VoIP termination service which routes
> the call to a PSTN number. Everything sounds great on the SIP phone, but
> the sound on the other end of the line is distant and missing bass, most
> especially so on the SPA-941 (which is the phone we really want to use).
> If I use the default handset mic gain value of 0db, the sound is so loud
> for the other person they have to hold the phone away from their ear. If I
> set it to -6db, it is still too quiet. The Aastra 9113i sounds a little
> better, and the Uniden 5.4 GHz Cordless sounds actually very good, so
I'm
> pretty sure my network setup is capable of transmitting good sound. Using
> the speaker-phone on the SPA-941 sounds significantly better than using
> the handset. But we need the handset to also sound good.
>
> I've tried different providers etc. and always come back to the phone.
I'm
> using G711u codec in all cases and silence suppresion is off.
>
> I saw a previous thread that mentioned changing the RTP from .03 to .02,
> however the post was regarding a MeetMe issue. I tried anyway, and it
> introduced an echo on the line.
>
> I've seen many rave reviews regarding the sound quality on the SPA-941,
so
> I'm wondering if maybe I got a bum handset? Would anyone be willing to
> receive/place a call to tell me if it sounds the way its supposed to or if
> there is indeed a problem?
>
> All suggestions/recommendations greatly appreciated.
>
> Much thanks,
>
> -- Ron
> ron@iamron.net
>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>

Andrew Joakimsen wrote:
> I too seem to have the same problem, dont know about "poor
quality"
> but its certainly not loud enough, I have to put my mouth to the
> microphone, otherwise the other end reports they cannot hear me. This
> does however seem to do a good job to cancel out the background noise
In the SIPura setup change the packet size from .3 to .2.