codec transcodation not working

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codec transcodation not working

neeraj.p
Hey,

I tried to establish audio call between two clients having different codec supports . I expected that transcoding  would take place.
But I am getting a codec negotiation error . 

Here is the information about client I am using 

leg A :
suppoerted codec : speex 

leg B:
supported codec : G722,PCMU,PCMA,GSM,OPUS


Here is some lines from my freeswitch logs

2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR.  SDP:
v=0
o=Zoiper 0 0 IN IP4 120.63.38.94
s=Zoiper
c=IN IP4 120.63.38.94
t=0 0
m=audio 11525 RTP/AVP 110
a=rtpmap:110 speex/8000

Here is some info about my freeswitch 

version - 1.4 beta

codecs supported - codec,ADPCM (IMA),mod_spandsp
codec,AMR,mod_amr
codec,B64 (STANDARD),mod_b64
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,G.729,mod_g729
codec,GSM,mod_spandsp
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_spandsp
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,VP8 Video (passthru),mod_vp8

My sip_profile configurations 
<param name="disable-transcoding" value="false"/>
<param name="inbound-late-negotiation" value="true"/>

I also tried with  inbound-late-negotiaion=false . But still getting codec negotiation error. 

Please help .



Regards,
Neeraj
 





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Re: codec transcodation not working

Michael Jerris
Administrator
can you post a pastebin link to a full log of this call?

On Oct 22, 2013, at 3:31 AM, "neeraj.p" <[hidden email]> wrote:

Hey,

I tried to establish audio call between two clients having different codec supports . I expected that transcoding  would take place.
But I am getting a codec negotiation error . 

Here is the information about client I am using 

leg A :
suppoerted codec : speex 

leg B:
supported codec : G722,PCMU,PCMA,GSM,OPUS


Here is some lines from my freeswitch logs

2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR.  SDP:
v=0
o=Zoiper 0 0 IN IP4 120.63.38.94
s=Zoiper
c=IN IP4 120.63.38.94
t=0 0
m=audio 11525 RTP/AVP 110
a=rtpmap:110 speex/8000

Here is some info about my freeswitch 

version - 1.4 beta

codecs supported - codec,ADPCM (IMA),mod_spandsp
codec,AMR,mod_amr
codec,B64 (STANDARD),mod_b64
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,G.729,mod_g729
codec,GSM,mod_spandsp
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_spandsp
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,VP8 Video (passthru),mod_vp8

My sip_profile configurations 
<param name="disable-transcoding" value="false"/>
<param name="inbound-late-negotiation" value="true"/>

I also tried with  inbound-late-negotiaion=false . But still getting codec negotiation error. 

Please help .



Regards,
Neeraj
 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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http://wiki.freeswitch.org
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FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

Official FreeSWITCH Sites
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http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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Re: codec transcodation not working

Ashwin Rath
Hi

Did your try forcing a codec(s) by setting the absolute_codec_string var ?

try something like this <action application="export" data="absolute_codec_string=G722"/>  and disable late negotiation.


On Tue, Oct 22, 2013 at 6:54 PM, Michael Jerris <[hidden email]> wrote:
can you post a pastebin link to a full log of this call?

On Oct 22, 2013, at 3:31 AM, "neeraj.p" <[hidden email]> wrote:

Hey,

I tried to establish audio call between two clients having different codec supports . I expected that transcoding  would take place.
But I am getting a codec negotiation error . 

Here is the information about client I am using 

leg A :
suppoerted codec : speex 

leg B:
supported codec : G722,PCMU,PCMA,GSM,OPUS


Here is some lines from my freeswitch logs

2013-10-22 10:31:12.795786 [INFO] switch_ivr_originate.c:1190 Sending early media
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[G722:9:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMU:0:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[PCMA:8:8000:20:64000]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:2880 Audio Codec Compare [speex:110:8000:20:0]/[GSM:3:8000:20:13200]
2013-10-22 10:31:12.795786 [DEBUG] switch_core_media.c:3064 No 2833 in SDP.  Disable 2833 dtmf and switch to INFO
2013-10-22 10:31:12.795786 [ERR] mod_sofia.c:2122 CODEC NEGOTIATION ERROR.  SDP:
v=0
o=Zoiper 0 0 IN IP4 120.63.38.94
s=Zoiper
c=IN IP4 120.63.38.94
t=0 0
m=audio 11525 RTP/AVP 110
a=rtpmap:110 speex/8000

Here is some info about my freeswitch 

version - 1.4 beta

codecs supported - codec,ADPCM (IMA),mod_spandsp
codec,AMR,mod_amr
codec,B64 (STANDARD),mod_b64
codec,G.711 alaw,CORE_PCM_MODULE
codec,G.711 ulaw,CORE_PCM_MODULE
codec,G.722,mod_spandsp
codec,G.723.1 6.3k,mod_g723_1
codec,G.726 16k,mod_spandsp
codec,G.726 16k (AAL2),mod_spandsp
codec,G.726 24k,mod_spandsp
codec,G.726 24k (AAL2),mod_spandsp
codec,G.726 32k,mod_spandsp
codec,G.726 32k (AAL2),mod_spandsp
codec,G.726 40k,mod_spandsp
codec,G.726 40k (AAL2),mod_spandsp
codec,G.729,mod_g729
codec,GSM,mod_spandsp
codec,H.261 Video (passthru),mod_h26x
codec,H.263 Video (passthru),mod_h26x
codec,H.263+ Video (passthru),mod_h26x
codec,H.263++ Video (passthru),mod_h26x
codec,H.264 Video (passthru),mod_h26x
codec,LPC-10,mod_spandsp
codec,PROXY PASS-THROUGH,CORE_PCM_MODULE
codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE
codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE
codec,Speex,mod_speex
codec,VP8 Video (passthru),mod_vp8

My sip_profile configurations 
<param name="disable-transcoding" value="false"/>
<param name="inbound-late-negotiation" value="true"/>

I also tried with  inbound-late-negotiaion=false . But still getting codec negotiation error. 

Please help .



Regards,
Neeraj
 




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
[hidden email]
http://www.freeswitchsolutions.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Official FreeSWITCH Sites
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--
Ashwin Kumar Rath

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Re: codec transcodation not working

neeraj.p
In reply to this post by neeraj.p
Here is the full log when leg A calls leg B 

Here is the log when leg B calls leg A

I can see  different errors in these two cases .

Regards,
Neeraj



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Re: codec transcodation not working

neeraj.p
In reply to this post by neeraj.p
Hey Ashwinrath,

Yes , I tried doing so.
It is  not working.

Regards,
Neeraj

_________________________________________________________________________
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Re: codec transcodation not working

Donny Hardyanto
In reply to this post by neeraj.p
From what I am understand that in freeswitch leg A dont call leg B. It bridges them.

So it suppose like this

client 1 -(leg a)-> FS ---(bridge)-- FS -(leg b)-> client 2.

So when client 1 call FS, the codec from FS dan client 1 must match (speex and telephony event). This will be leg A.

then FS call client 2, the codec from FS and client 2 must match (OPUS for example and telephony event). This will be leg B.

Then you bridge them leg A to leg B. Then you have transcoding.

May be if you can show the corresponding dialplan we will know better.

Donny



On Tue, Oct 22, 2013 at 9:23 PM, neeraj.p <[hidden email]> wrote:
Here is the full log when leg A calls leg B 

Here is the log when leg B calls leg A

I can see  different errors in these two cases .

Regards,
Neeraj



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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_________________________________________________________________________
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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Re: codec transcodation not working

neeraj.p
In reply to this post by neeraj.p
Hey Donny 

Here is my Dialplan

--Neeraj

_________________________________________________________________________
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