[asterisk-users] Codec preference

[jonas kellens]

How can I set the prefered codec between 2 calling parties ??

My Grandstream supports G729, alaw and gsm... in this order.
The Zoiper softphone has alaw and gsm as codecs... in that order.

Although there should be a matching codec found, my Grandstream can not
call the Zoiper softphone.

CLI shows :

[Mar 11 17:47:21] WARNING[22367]: channel.c:3340
ast_channel_make_compatible: No path to translate from
SIP/mygrandstream-09c599e0(2) to SIP/zoiper-09cd57f8(256)
[Mar 11 17:47:21]     -- Got SIP response 415 "Unsupported Media Type"
back from 192.168.1.106 (<-- zoiper)

SIP debug :

[Mar 11 17:55:57] Peer audio RTP is at port 192.168.1.101:10110 (<-- the
Grandstream)
[Mar 11 17:55:57] Found audio description format PCMA for ID 8
[Mar 11 17:55:57] Found audio description format GSM for ID 3
[Mar 11 17:55:57] Found audio description format PCMU for ID 0
[Mar 11 17:55:57] Found audio description format G729 for ID 18
[Mar 11 17:55:57] Found audio description format telephone-event for ID
101
[Mar 11 17:55:57] Capabilities: us - 0x10a (gsm|alaw|g729), peer -
audio=0x10e (gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10a
(gsm|alaw|g729)
[Mar 11 17:55:57] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
...
[Mar 11 17:55:57] Audio is at 192.168.1.150 port 11586 (<-- my Asterisk)
[Mar 11 17:55:57] Adding codec 0x100 (g729) to SDP
[Mar 11 17:55:57] Adding non-codec 0x1 (telephone-event) to SDP

This is what Asterisk sends to the Zoiper in the INVITE (sdp) :

Content-Type: application/sdp
Content-Length: 263
v=0
o=root 3208 3208 IN IP4 192.168.1.150
s=session
c=IN IP4 192.168.1.150
t=0 0
m=audio 11586 RTP/AVP 18 101
a=rtpmap:18 G729/8000

Why isn't Asterisk negotiating with the Zoiper for the alaw-codec ??

The sip-configuration (realtime MySQL) for the Grandstream is :

allow : g729;alaw;gsm

and the Zoiper softphone :

allow : alaw;gsm;g729


Kind regards,

Jonas.
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