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 :) On Mon, Feb 15, 2016 at 10:25 AM, Ken Rice <krice@freeswitch.org <mailto:krice@freeswitch.org> > wrote: 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> [mailto: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 <mailto: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 <mailto:mike@jerris.com> > wrote: 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 <mailto:colton.conor@gmail.com> > wrote: 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 <mailto:krice@freeswitch.org> > wrote: 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
On Feb 14, 2016, at 3:37 PM, Rajil Saraswat <rajil.s@gmail.com <mailto: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?
On 14 February 2016 at 13:56, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca> > wrote: fyi https://freeswitch.org/confluence/display/FREESWITCH/Codec+Negotiation
On Sun, Feb 14, 2016 at 2:04 PM, Rajil Saraswat <rajil.s@gmail.com <mailto: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 <mailto:gmaruzz@gmail.com> > wrote:
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 <mailto:rajil.s@gmail.com> > ha scritto:
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 <mailto: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 <mailto: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 <mailto:sofia/internal/303@192.168.1.5> [CS_CONSUME_MEDIA] [SERVICE_NOT_IMPLEMENTED]
Thanks Rajil
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