[asterisk-users] Codec preference

[jonas kellens]

If I have this is sip.conf :

[general]
disallow=all
allow=g729
allow=alaw

The prefered codecs set in my Grandstream phone is G729, alaw.

In the sip peer definition I have commented out 'disallow=' and
'allow='.

The prefered codecs set in the Zoiper softphone is alaw, gsm.

In the sip peer definition I have commented out 'disallow=' and
'allow='.

When making a call from the Grandstream to the Zoiper softphone, with
Asterisk staying in the media path (canreinvite=no), you would expect
all 3 of them to use alaw.
But this is what happens :

The Grandstream :
v=0
o=test3 8000 8001 IN IP4 192.168.1.101 (<-- Grandstream IP-address)
s=SIP Call
c=IN IP4 192.168.1.101
t=0 0
m=audio 10082 RTP/AVP 18 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

[Mar 12 11:44:14] Found RTP audio format 18
[Mar 12 11:44:14] Found RTP audio format 8
[Mar 12 11:44:14] Found RTP audio format 101
[Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082
[Mar 12 11:44:14] Found audio description format G729 for ID 18
[Mar 12 11:44:14] Found audio description format PCMA for ID 8
[Mar 12 11:44:14] Found audio description format telephone-event for ID
101
[Mar 12 11:44:14] Capabilities: us - 0x108 (alaw|g729), peer -
audio=0x108 (alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|
g729)
[Mar 12 11:44:14] Non-codec capabilities (dtmf): us - 0x1
(telephone-event), peer - 0x1 (telephone-event), combined - 0x1
(telephone-event)
[Mar 12 11:44:14] Peer audio RTP is at port 192.168.1.101:10082

[Mar 12 11:44:14]     -- Called test1 (<-- the Zoiper softphone)
[Mar 12 11:44:14]     -- Got SIP response 415 "Unsupported Media Type"
back from 192.168.1.106 (IP-address of Zoiper softphone)


So what happens here with the codec negotiation between Asterisk and the
Zoiper softphone ??

Making the call the other way around (Zoiper calls Grandstream) the call
succeeds and the codec is alaw... like it should be.
I do get the warnings :
[Mar 12 11:53:30] WARNING[23703]: channel.c:3340
ast_channel_make_compatible: No path to translate from
SIP/test3-0a168b48(256) to SIP/test1-0a166d00(8)
[Mar 12 11:53:31]     -- SIP/test3-0a168b48 is ringing
[Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to
find a codec translation path from 0x8 (alaw) to 0x100 (g729)
[Mar 12 11:53:34] WARNING[23670]: channel.c:2961 set_format: Unable to
find a codec translation path from 0x8 (alaw) to 0x100 (g729)
[Mar 12 11:53:34]     -- SIP/test3-0a168b48 answered SIP/test1-0a166d00
[Mar 12 11:53:34]     -- Packet2Packet bridging SIP/test1-0a166d00 and
SIP/test3-0a168b48

(I know there are G729-licences to translate from G729 to alaw, but if
both support alaw, then alaw should be the negotiated codec, no ?!)

Can this be explained ?


Thanks.
Jonas.
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