We've bought the G729 codec for lowering SIP bandwidth usage (we're using grandstream phones) and we're quite happy with it up until I tried using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. Weirdly enough, calls from IAXphone to the GS phone work just fine. So are calls from both phones to voicemail. And from both phones to analog phones connected to FXS ports. Calls from GS to IAXphone ring, and once I answer the call in IAXphone, I hear a very load noise. Asterisk CLI shows this: channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/205/1 of format GSM since our native format has changed to G729A (not just once, over and over and over again till I hang up) my sip.conf entry for the grandstream phone shows disallow=all allow=g729 and reinvite=no I did 'iax2 show channels' and 'sip show channels' When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and the Sip channel shows codec G729A When I call the other way around, Sip shows G729A and IAX2 shows GSM. Hmm, seems ok... I tried changing my sip conf to include allow=g729,gsm Now the calls sounds fine, but the bandwidth is uses is near 20K instead of just 6K (both phones are near me, and the Asterisk server is at a remote location, and I can monitor bandwidth usage in my FW). Can anyone help? Thanks. Shoval Tomer, IT Manager, SofTov Advanced Systems, Ltd. Office: +972-3-9230686 ext. 179 Fax: +972-3-9216642 Mobile: +972-54-8000200
Shoval Tomer wrote:>We've bought the G729 codec for lowering SIP bandwidth usage (we're >using grandstream phones) and we're quite happy with it up until I tried >using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. > >Weirdly enough, calls from IAXphone to the GS phone work just fine. >So are calls from both phones to voicemail. And from both phones to >analog phones connected to FXS ports. > >Calls from GS to IAXphone ring, and once I answer the call in IAXphone, >I hear a very load noise. > >On which side do you hear this noise? IAXphone, or GS? Does the noise continue, or just go for a few milliseconds? Does your version of IAXphone support multiple codecs, or just GSM?>Asterisk CLI shows this: >channel.c:1314 ast_read: Dropping incompatible voice frame on IAX2/205/1 >of format GSM since our native format has changed to G729A > >(not just once, over and over and over again till I hang up) > >my sip.conf entry for the grandstream phone shows >disallow=all >allow=g729 >and >reinvite=no > >I did 'iax2 show channels' and 'sip show channels' > >When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and >the Sip channel shows codec G729A > >When I call the other way around, Sip shows G729A and IAX2 shows GSM. > >Hmm, seems ok... > >I tried changing my sip conf to include allow=g729,gsm > >Now the calls sounds fine, but the bandwidth is uses is near 20K instead >of just 6K (both phones are near me, and the Asterisk server is at a >remote location, and I can monitor bandwidth usage in my FW). > >Can anyone help? > >It obviosuly sounds like codec negotion, on one side or another, isn't working, and you're sending an incompatible codec to the other side, or the other side doesn't know what codec is being sent.. Using ethereal to see what's happening on the network would show you what's going on pretty clearly.. -SteveK
Thanks Steve, See my answers inline> -----Original Message----- > From: Steve Kann [mailto:stevek@stevek.com] > Sent: Tuesday, December 21, 2004 1:49 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] codec issues > > Shoval Tomer wrote: > > >We've bought the G729 codec for lowering SIP bandwidth usage (we're > >using grandstream phones) and we're quite happy with it up until Itried> >using IAXPhone 0.2.0 build 116 with my asterisk 1.0.0 installations. > > > >Weirdly enough, calls from IAXphone to the GS phone work just fine. > >So are calls from both phones to voicemail. And from both phones to > >analog phones connected to FXS ports. > > > >Calls from GS to IAXphone ring, and once I answer the call inIAXphone,> >I hear a very load noise. > > > > > On which side do you hear this noise? IAXphone, or GS? >I hear the noise on IAX site> Does the noise continue, or just go for a few milliseconds?Continue till I hang up> > Does your version of IAXphone support multiple codecs, or just GSM?Supports only GSM> > >Asterisk CLI shows this: > >channel.c:1314 ast_read: Dropping incompatible voice frame onIAX2/205/1> >of format GSM since our native format has changed to G729A > > > >(not just once, over and over and over again till I hang up) > > > >my sip.conf entry for the grandstream phone shows > >disallow=all > >allow=g729 > >and > >reinvite=no > > > >I did 'iax2 show channels' and 'sip show channels' > > > >When I call from IAXPhone to GS, the IAX2 channel shows codec GSM and > >the Sip channel shows codec G729A > > > >When I call the other way around, Sip shows G729A and IAX2 shows GSM. > > > >Hmm, seems ok... > > > >I tried changing my sip conf to include allow=g729,gsm > > > >Now the calls sounds fine, but the bandwidth is uses is near 20Kinstead> >of just 6K (both phones are near me, and the Asterisk server is at a > >remote location, and I can monitor bandwidth usage in my FW). > > > >Can anyone help? > > > > > > It obviosuly sounds like codec negotion, on one side or another, isn't > working, and you're sending an incompatible codec to the other side,or> the other side doesn't know what codec is being sent.. >How can I even control that? And sip show channels and iax2 show channels show the correct codecs.> Using ethereal to see what's happening on the network would show you > what's going on pretty clearly.. > > -SteveK >I'll try but it's not going to be easily done, as the asterisk server is at a remote location and I'm no ethereal expert...> > > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > This message has been scanned for viruses and > dangerous content by MailScanner, and is > believed to be clean. > MailScanner thanks transtec Computers for their support.
On Tue, 21 Dec 2004 01:10:51 +0200, Shoval Tomer <shoval@softov.co.il> wrote:> Now the calls sounds fine, but the bandwidth is uses is near 20K instead > of just 6K (both phones are near me, and the Asterisk server is at a > remote location, and I can monitor bandwidth usage in my FW).I don't think you'll ever really see just 6 Kbps on the wire as IP, UDP, and RTP overhead for a 30 ms sample is around 10 Kbps all by itself. On an 8 Kbps codec like G.729 it wouldn't be unusual to see 18 Kbps of bandwidth used by the conversation. If it were using GSM you should see about 29 to 30 Kbps as it is a 13 Kbps codec plus 50 packets worth of overhead, 16 Kbps. (20 ms samples) I don't know why you're not seeing the same codecs indicated on both sides but it looks to me like you are getting G.729.