No compatible codecs!


#1

Trying to use G.726 codec, i receive this error messages:

… No compatible codecs!

I enabled SIP DEBUG on the Asterisk console
and receive this :

By the way i am running FC4.

Anybody have expenrience to share with codec g726 or speex wideband ?

*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:

0 headers, 0 lines
Urgent handler

Sip read:
INVITE sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-;rport
Max-Forwards: 70
Contact: sip:3960@10.34.6.70:9006
To: sip:5555@canamdc.ws
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 222

v=0
o=- 4383112 4383130 IN IP4 10.34.6.70
s=eyeBeam
c=IN IP4 10.34.6.70
t=0 0
m=audio 9008 RTP/AVP 5 101
a=alt:1 1 : FF590EAE 00000014 10.34.6.70 9008
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

13 headers, 10 lines
Using latest request as basis request
Sending to 10.34.6.70 : 9006 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
To: sip:5555@canamdc.ws;tag=as532e4a6e
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:5555@10.34.6.3
Proxy-Authenticate: Digest realm=“asterisk”, nonce="4ef73b6c"
Content-Length: 0

to 10.34.6.70:9006
Scheduling destruction of call ‘4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…’ in 15000 ms
Found user '3960’
Urgent handler

Sip read:
ACK sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-826eb02d9f437e7d-1–d87543-;rport
To: sip:5555@canamdc.ws;tag=as532e4a6e
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 1 ACK
Content-Length: 0

7 headers, 0 lines
Urgent handler

Sip read:
INVITE sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-;rport
Max-Forwards: 70
Contact: sip:3960@10.34.6.70:9006
To: sip:5555@canamdc.ws
From: “Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username=“3960”,realm=“asterisk”,nonce=“4ef73b6c”,uri="sip:5555@canamdc.ws”,response=“f131392a9be2324d6cbc2461ff77de78”,algorithm=MD5
Supported: eventlist
User-Agent: eyeBeam release 3010n stamp 19039
Content-Length: 222

v=0
o=- 4383112 4383130 IN IP4 10.34.6.70
s=eyeBeam
c=IN IP4 10.34.6.70
t=0 0
m=audio 9008 RTP/AVP 5 101
a=alt:1 1 : FF590EAE 00000014 10.34.6.70 9008
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv

14 headers, 10 lines
Using latest request as basis request
Sending to 10.34.6.70 : 9006 (non-NAT)
Found user '3960’
Found RTP audio format 5
Found RTP audio format 101
Peer audio RTP is at port 10.34.6.70:9008
Found description format telephone-event
Capabilities: us - 0x10 (g726), peer - audio=0x20 (adpcm)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 0x1 (g723)
Nov 15 13:32:49 NOTICE[4406]: chan_sip.c:2792 process_sdp: No compatible codecs!Transmitting (no NAT):
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
To: sip:5555@canamdc.ws;tag=as532e4a6e
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:5555@10.34.6.3
Content-Length: 0

to 10.34.6.70:9006
Destroying call '4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…'
Urgent handler

Sip read:
ACK sip:5555@canamdc.ws SIP/2.0
Via: SIP/2.0/UDP 10.34.6.70:9006;branch=z9hG4bK-d87543-3f64b826b6074a54-1–d87543-;rport
To: sip:5555@canamdc.ws;tag=as532e4a6e
From: "Eric Talbot"sip:3960@canamdc.ws;tag=4f2b3b33
Call-ID: 4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…
CSeq: 2 ACK
Content-Length: 0

7 headers, 0 lines
Destroying call '4414295847532a52@Y21pMzM4MTUuY2FuYW1kYy53cw…'
Urgent handler :exclamation:


#2

[quote=“ciretoblat”]Trying to use G.726 codec, i receive this error messages:

… No compatible codecs!
[/quote]
Have you installed a g726 codec? Asterisk doesn’t come with it installed. You’ll no doubt need to buy a licence and install it yourself.


#3

Really thought g726 was for free. As specified in OREILLY’s “ASTERISK THE FUTUR OF TELEPHONY” Chapiter 8 , p145 says license is not required.

Therefore, can you tell me where can i get it, otherwise can you tell me if there’s another codec apart g711a or g711u giving very good results on high bandwith with Asterisk ?


#4

Well that’s probably true then. But it’s not shipped with Asterisk.

No idea, sorry. But what’s wrong with g711? It’s the industry standard.


#5

Nothing is really wrong with it. Except that 726 uses less bandwidth and in the environment where i am, it may be very interesting.

Thanks for you help anyway.