[Freeswitch-users] Codec Mismatch Because of Different Frame Size

Thu Sep 30 14:09:35 PDT 2010

  I checked out and compiled the current git version some hours ago. 
Unfortunately the problem seems to be still not fixed. I tried to 
examine it more detailed. I will attach this e-mail additionally in a 
text file to avoid that it get unreadable by line wrapping.

A call from a Cisco is working, if "codec g729 bytes 60" is configured only:

v=0
o=CiscoSystemsSIP-GW-UserAgent 5025 3066 IN IP4 a.b.c.d
s=SIP Call
c=IN IP4 a.b.c.d
t=0 0
m=audio 16408 RTP/AVP 18
c=IN IP4 a.b.c.d
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:60

2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:3873 Activate Buggy 
RFC2833 Mode!
m2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:4229 Audio Codec 
Compare [G729:18:8000:60:0]/[G729:18:8000:20:8000]
m2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:4282 Substituting codec 
G729 at 60i@8000h
2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:3873 Activate Buggy 
RFC2833 Mode!
2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:60:0]/[G729:18:8000:20:8000]
2010-09-30 21:37:44.297076 [DEBUG] sofia_glue.c:4282 Substituting codec 
G729 at 60i@8000h

For a Cisco, on which I configured a codec list with frame sizes of 80 
bytes, the matching is working as well. Maybe it's because no ptime is 
set at all:

2010-09-30 22:06:45.598963 [DEBUG] sofia.c:4469 Channel 
sofia/external/3039731964 at a.b.c.d entering state [received][100]
2010-09-30 22:06:45.598963 [DEBUG] sofia.c:4480 Remote SDP:
v=0
o=CiscoSystemsSIP-GW-UserAgent 6897 7698 IN IP4 a.b.c.d
s=SIP Call
c=IN IP4 a.b.c.d
t=0 0
m=audio 17640 RTP/AVP 18 8 0
c=IN IP4 a.b.c.d
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000

2010-09-30 22:06:45.598963 [DEBUG] sofia_glue.c:3873 Activate Buggy 
RFC2833 Mode!
2010-09-30 22:06:45.598963 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:20:0]/[G729:18:8000:20:8000]
2010-09-30 22:06:45.598963 [DEBUG] sofia_glue.c:2600 Set Codec 
sofia/external/3039731964 at a.b.c.d G729/8000 20 ms 160 samples 8000 bits

But I still can observe problems, when a codec list is used and some 
kind of ptime is set. Unfortunately I can't trigger such calls, I have 
to wait until I receive them from my customers:

2010-09-30 22:50:14.152164 [NOTICE] switch_channel.c:779 New Channel 
sofia/external/12345 at a.b.c.d [c02c759a-2d43-445f-992f-8c3f59d6171a]
2010-09-30 22:50:14.152164 [DEBUG] sofia.c:4469 Channel 
sofia/external/12345 at a.b.c.d entering state [received][100]
2010-09-30 22:50:14.152164 [DEBUG] sofia.c:4480 Remote SDP:
v=0
o=Sonus_UAC 10702 20259 IN IP4 a.b.c.d
s=SIP Media Capabilities
c=IN IP4 195.219.240.20
t=0 0
m=audio 26128 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=maxptime:30

2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[G729:18:8000:20:8000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[G729:18:8000:20:8000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[G729:18:8000:20:8000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[G729:18:8000:20:8000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[G729:18:8000:20:8000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[PCMU:0:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[PCMU:0:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[PCMU:0:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[PCMU:0:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[PCMU:0:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[PCMA:8:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[PCMA:8:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[PCMA:8:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[PCMA:8:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[PCMA:8:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[G7221:115:32000:20:48000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[G7221:115:32000:20:48000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[G7221:115:32000:20:48000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[G7221:115:32000:20:48000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[G7221:115:32000:20:48000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[G7221:107:16000:20:32000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[G7221:107:16000:20:32000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[G7221:107:16000:20:32000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[G7221:107:16000:20:32000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[G7221:107:16000:20:32000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[G722:9:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[G722:9:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[G722:9:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[G722:9:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[G722:9:8000:20:64000]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMA:8:8000:30:0]/[GSM:3:8000:20:13200]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[PCMU:0:8000:30:0]/[GSM:3:8000:20:13200]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G729:18:8000:30:0]/[GSM:3:8000:20:13200]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[G723:4:8000:30:0]/[GSM:3:8000:20:13200]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4229 Audio Codec Compare 
[telephone-event:101:8000:30:0]/[GSM:3:8000:20:13200]
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] sofia_glue.c:4331 Set 2833 dtmf 
send/recv payload to 101
2010-09-30 22:50:14.152164 [DEBUG] switch_channel.c:2359 
(sofia/external/12345 at a.b.c.d) Callstate Change DOWN -> HANGUP
2010-09-30 22:50:14.152164 [NOTICE] sofia.c:4686 Hangup 
sofia/external/12345 at a.b.c.d [CS_NEW] [INCOMPATIBLE_DESTINATION]

Does anybody have any ideas?

Thank you in advance
               Jan

Am 30.09.2010 18:46, schrieb Brian West:
> Codec interface was changed a little bit yesterday to support per codec fmtp parsing and bitrates for g722.1 which was the reason it was changed.  Please make sure your tree is clean and do a make current again and see if that solves your problem.
>
> /b
>
> On Sep 30, 2010, at 11:24 AM, Brock Williams wrote:
>
>> We are experiencing the same problem on a git pull from yesterday.
>>
>> Brock
>>
>> On 09/30/2010 05:16 AM, Jan Riedinger wrote:
>>>   It's the git version from yesterday evening:
>>>
>>>     freeswitch at internal>  version
>>>     FreeSWITCH Version 1.0.head (git-633f193 2010-09-29 16-52-43 -0500)
>>>
>>> BR
>>>      Jan
>>>
>
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-- 
Jan Riedinger                           Phone :  +49-30-39 73 19 66
Dipl.-Inf. | Managing Director          Fax   :  +49-30-39 73 19 64
                                         E-Mail:  riedinger at sns.eu
SNS Consult GmbH                        ICQ   :  163-237-041
Südwestkorso 49a                        MSN   :  jan at sns-consult.de
14197 Berlin GERMANY                    Skype :  Jan Riedinger

AG Charlottenburg - HRB 71973

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