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rajil.s at gmail.com
Guest





PostPosted: Sat Feb 13, 2016 9:55 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0

------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0

------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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gmaruzz at gmail.com
Guest





PostPosted: Sun Feb 14, 2016 12:30 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

How you originate the call? Is a bridge? From which phone?
Also, please pastebin the complete sip trace (from start of leg A to end of both legs) and put here a link to pastebin
Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:
Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:
   ------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP
192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0

   ------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:
   ------------------------------------------------------------------------
   ACK sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email]) SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0

   ------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 (303@192.168.1.5) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 (303@192.168.1.5) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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rajil.s at gmail.com
Guest





PostPosted: Sun Feb 14, 2016 2:06 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com> wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:

------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP

192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0


------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:

------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0


------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
rtreleaven at bunnykic...
Guest





PostPosted: Sun Feb 14, 2016 2:35 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

Look at the invite starting at 480. You are only offering opus to the callee. On Feb 14, 2016 2:05 PM, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:
The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)> wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:

------------------------------------------------------------------------
    SIP/2.0 406 Not Acceptable
    Via: SIP/2.0/UDP

192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
    From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
    To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
    CSeq: 87372504 INVITE
    Content-Length:  0


------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:

------------------------------------------------------------------------
    ACK sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email]) SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
    Max-Forwards: 68
    From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
    To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
    CSeq: 87372504 ACK
    Content-Length: 0


------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 (303@192.168.1.5) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 (303@192.168.1.5) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
Back to top
rtreleaven at bunnykic...
Guest





PostPosted: Sun Feb 14, 2016 2:58 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:
Quote:
The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)> wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:

------------------------------------------------------------------------
    SIP/2.0 406 Not Acceptable
    Via: SIP/2.0/UDP

192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
    From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
    To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
    CSeq: 87372504 INVITE
    Content-Length:  0


------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:

------------------------------------------------------------------------
    ACK sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email]) SIP/2.0
    Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
    Max-Forwards: 68
    From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
    To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
    Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
    CSeq: 87372504 ACK
    Content-Length: 0


------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 (303@192.168.1.5) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 (303@192.168.1.5) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


Back to top
rajil.s at gmail.com
Guest





PostPosted: Sun Feb 14, 2016 4:39 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
Quote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Quote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


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krice at freeswitch.org
Guest





PostPosted: Sun Feb 14, 2016 5:08 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
ACK sip:303@192.168.1.5 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


_________________________________________________________________________
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consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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colton.conor at gmail.com
Guest





PostPosted: Sun Feb 14, 2016 7:22 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.

Plus if you use TLS for encryption and security then you are already using TCP right? 


On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:
This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically.  I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)>
wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
   ACK sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email]) SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <sip:202@192.168.1.111 ([email]sip%3A202@192.168.1.111[/email])>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5 ([email]sip%3A303@192.168.1.5[/email])>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 (303@192.168.1.5) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 (303@192.168.1.5) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


_________________________________________________________________________
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consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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mike at jerris.com
Guest





PostPosted: Mon Feb 15, 2016 10:35 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

any device that even remotely follows sipspecs supports TCP.  Most phones I have seen do.

On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:
So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.

Plus if you use TLS for encryption and security then you are already using TCP right? 


On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <[url=javascript:_e(%7B%7D,'cvml','krice@freeswitch.org');]krice@freeswitch.org[/url]> wrote:
Quote:
This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically.  I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <[url=javascript:_e(%7B%7D,'cvml','rtreleaven@bunnykick.ca');]rtreleaven@bunnykick.ca[/url]> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <[url=javascript:_e(%7B%7D,'cvml','gmaruzz@gmail.com');]gmaruzz@gmail.com[/url]>
wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <[url=javascript:_e(%7B%7D,'cvml','rajil.s@gmail.com');]rajil.s@gmail.com[/url]> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url]
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <[url=javascript:_e(%7B%7D,'cvml','sip:202@192.168.1.111');]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
   To: <[url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
   ACK [url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url] SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <[url=javascript:_e(%7B%7D,'cvml','sip:202@192.168.1.111');]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
   To: <[url=javascript:_e(%7B%7D,'cvml','sip:303@192.168.1.5');]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url] entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/[url=javascript:_e(%7B%7D,'cvml','303@192.168.1.5');]303@192.168.1.5[/url] [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


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colton.conor at gmail.com
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PostPosted: Mon Feb 15, 2016 11:06 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 

On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote:
any device that even remotely follows sipspecs supports TCP.  Most phones I have seen do.

On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:
So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.

Plus if you use TLS for encryption and security then you are already using TCP right? 


On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org> wrote:
Quote:
This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically.  I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com>
wrote:
Quote:
How you originate the call? Is a bridge? From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
   ACK sip:303@192.168.1.5 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


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mike at jerris.com
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PostPosted: Mon Feb 15, 2016 11:10 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

It is heavier but I think that otherwise is superior.
Quote:
On Feb 15, 2016, at 11:04 AM, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.
On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote:
any device that even remotely follows sipspecs supports TCP. Most phones I have seen do.On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:
So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
Plus if you use TLS for encryption and security then you are already using TCP right?

On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation >> >>> On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <http://pastebin.com/xiGqtj1Y >>> >>> The call is being made from 303 (Android/CSipsimple with OPUS codec) >>> to 208 (pjsua test client with PCMU codec). The error is on line 545. >>> >>> On 14 February 2016 at 11:28, Giovanni Maruzzelli <http://www.freeswitchsolutions.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://confluence.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> http://www.freeswitchsolutions.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://confluence.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org






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krice at freeswitch.org
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PostPosted: Mon Feb 15, 2016 11:27 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

The problem isn’t necessarily the devices, but there is also the carriers…  

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.


On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote:

any device that even remotely follows sip
specs supports TCP. Most phones I have seen do.


On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.


Plus if you use TLS for encryption and security then you are already using TCP right?



On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)>
wrote:
Quote:
How you originate the call? Is a bridge? >From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <[url=sip:202@192.168.1.111]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
ACK [url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <[url=sip:202@192.168.1.111]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


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colton.conor at gmail.com
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PostPosted: Mon Feb 15, 2016 11:31 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

True,

But freeswitch talking to the carriers is almost always UPD.


However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too Smile


On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:

The problem isn’t necessarily the devices, but there is also the carriers…  
 
From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode
 
So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior. 

 
On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote:

any device that even remotely follows sip
specs supports TCP.  Most phones I have seen do.


On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.
 

Plus if you use TLS for encryption and security then you are already using TCP right? 


 
On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically.  I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)>
wrote:
Quote:
How you originate the call? Is a bridge? >From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as  global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
   SIP/2.0 406 Not Acceptable
   Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   CSeq: 87372504 INVITE
   Content-Length:  0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
   ACK sip:303@192.168.1.5 SIP/2.0
   Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
   Max-Forwards: 68
   From: "202" <sip:202@192.168.1.111>;tag=DFX0FUvr2vNcm
   To: <sip:303@192.168.1.5>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
   Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
   CSeq: 87372504 ACK
   Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


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krice at freeswitch.org
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PostPosted: Mon Feb 15, 2016 11:37 am    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

The problem still exists for expanding SDPs… using TCP to the user/device then trying to send the same thing out to the carrier over UDP is what was causing the problem in the first place… so the decision was made to prevent those problems we’ll only offer what the device offers and not expand the number of codecs even further increasing the already bloated SDPs to the point where they fragment over UDP and get dropped…

So is TCP better for some things, yes it is, however, the lack of market wide support for it with carriers makes it a pain in the ass even tho the RFCs specifically say you MUST support both UDP and TCP for SIP, but certain VoIP softwares out there only implemented UDP many years ago and now we’re stuck with that legacy

From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:30 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

True,


But freeswitch talking to the carriers is almost always UPD.



However, freeswitch talking to the clients I would say TCP would be idea. So its almost like freeswitch is trancoding from TCP to UDP too Smile



On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:

The problem isn’t necessarily the devices, but there is also the carriers…

From: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] On Behalf Of Colton Conor
Sent: Monday, February 15, 2016 10:04 AM
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)>
Subject: Re: [Freeswitch-users] Freeswitch doesnt transcode

So if the device supports TCP, is there any reason not to use TCP. AKA is there any reason to keep on using UDP. TCP seems superior.


On Mon, Feb 15, 2016 at 9:33 AM, Michael Jerris <mike@jerris.com (mike@jerris.com)> wrote:
Quote:

any device that even remotely follows sip
specs supports TCP. Most phones I have seen do.


On Sunday, February 14, 2016, Colton Conor <colton.conor@gmail.com (colton.conor@gmail.com)> wrote:
Quote:

So is TCP the preferred method of doing SIP these days? I like TCP with endpoints as they always break through firewalls and we never seem to have in issue with TCP. However UDP is a headache. So if you have the choice why not do TCP? I realize some devices only support UDP, but the majority of SIP phones out there today do support TCP.


Plus if you use TLS for encryption and security then you are already using TCP right?



On Sun, Feb 14, 2016 at 4:07 PM, Ken Rice <krice@freeswitch.org (krice@freeswitch.org)> wrote:
Quote:

This behavior changed a while ago. This was dictates by ever growing SDPs and exceeding MTUs causing udp fragmentation. Udp does not deal with fragmentation and everyone refuses to fully implement sip over tcp for some reason even tho a ton of things support it and the RFCs require it

Sent from my iPhone

Quote:
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

Thanks, after setting media_mix_inbound_outbound_codecs=true,
transcoding happens automatically. I remember not setting this
variable in other installations and transcoding used to work out of
the box. Is media_mix_inbound_outbound_codecs=true default in
Freeswitch?

Quote:
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation

Quote:
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com (rajil.s@gmail.com)> wrote:

The siptrace is at http://pastebin.com/xiGqtj1Y

The call is being made from 303 (Android/CSipsimple with OPUS codec)
to 208 (pjsua test client with PCMU codec). The error is on line 545.

On 14 February 2016 at 11:28, Giovanni Maruzzelli <gmaruzz@gmail.com (gmaruzz@gmail.com)>
wrote:
Quote:
How you originate the call? Is a bridge? >From which phone?

Also, please pastebin the complete sip trace (from start of leg A to end
of
both legs) and put here a link to pastebin

Il 14/Feb/2016 03:54, "Rajil Saraswat" <rajil.s@gmail.com (rajil.s@gmail.com)> ha scritto:
Quote:

Hello,

I have a remote sip phone (Linksys SPA3102) which only supports PCMU.
When I call to this remote sip phone i get a 406 error that opus is
not supported as shown by the sip trace below. However, if I force the
codec to absolute like this
{absolute_codec_string='PCMU,PCMA'}sofia/internal/303@192.168.1.5 (303@192.168.1.5)
the call works fine.

Is there anyway I can make FreeSWITCH to automatically transcode
without forcing the codec string in the dial plan?

The codec preferences is set as global_codec_prefs=OPUS,PCMU,PCMA and
outbound_codec_prefs=PCMU,PCMA,GSM

---------------------------siptrace--------------------------------

recv 333 bytes from udp/[192.168.1.5]:5060 at 08:02:16.368499:


------------------------------------------------------------------------
SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP


192.168.1.111;rport=5060;received=192.168.1.111;branch=z9hG4bKeS356tttajjej
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
From: "202" <[url=sip:202@192.168.1.111]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
CSeq: 87372504 INVITE
Content-Length: 0



------------------------------------------------------------------------
send 324 bytes to udp/[192.168.1.5]:5060 at 08:02:16.368591:


------------------------------------------------------------------------
ACK [url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.111;rport;branch=z9hG4bKeS356tttajjej
Max-Forwards: 68
From: "202" <[url=sip:202@192.168.1.111]sip:202@192.168.1.111[/url]>;tag=DFX0FUvr2vNcm
To: <[url=sip:303@192.168.1.5]sip:303@192.168.1.5[/url]>;tag=htMF9ckdglw3EJRZaILd6XWt4uVKAO8q
Call-ID: 00ff246a-4d66-1234-f4b2-74d02b7a3124
CSeq: 87372504 ACK
Content-Length: 0



------------------------------------------------------------------------
2016-02-14 08:02:16.356283 [DEBUG] sofia.c:6760 Channel
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) entering state [terminated][406]
2016-02-14 08:02:16.356283 [NOTICE] sofia.c:7779 Hangup
sofia/internal/303@192.168.1.5 ([email]sofia/internal/303@192.168.1.5[/email]) [CS_CONSUME_MEDIA]
[SERVICE_NOT_IMPLEMENTED]

Thanks
Rajil


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

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_________________________________________________________________________
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consulting@freeswitch.org (consulting@freeswitch.org)
http://www.freeswitchsolutions.com

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

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PostPosted: Mon Feb 15, 2016 1:28 pm    Post subject: [Freeswitch-users] Freeswitch doesnt transcode Reply with quote

On Sun, Feb 14, 2016 at 1:56 PM, Russell Treleaven <rtreleaven@bunnykick.ca (rtreleaven@bunnykick.ca)> wrote:
Quote:
fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation






That is a really lucid and complete description of CODEC negotiation in FS.  Thanks for the link, and thanks to the original author for writing it.


Bob
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