[Freeswitch-users] Number of codecs offerred in SDP
Mark Campbell-Smith
mcampbellsmith at gmail.com
Wed Jun 30 06:07:32 PDT 2010
Updating the configuration did not help.
I'm not telling FS to use any other codecs other than what I have
specified below, so I'm not sure what I have done wrong. Below is the
full trace:
------------------------------------------------------------------------
INVITE sip:1020 at 192.168.1.120 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
From: 1000 <sip:1000 at 192.168.1.120>;tag=2c7a518d12f9370eo0
To: <sip:1020 at 192.168.1.120>
Call-ID: 316156b9-f6b413d2 at 192.168.1.121
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="1000",realm="192.168.1.120",nonce="77836a7a-8447-11df-93a6-d9ad5b204ca2",uri="sip:1020 at 192.168.1.120",algorithm=MD5,response="c4a5e5cb60676366b8bcfb1329f3fc08",qop=auth,nc=00000001,cnonce="6a0220d0"
Contact: 1000 <sip:1000 at 192.168.1.121:5060>
Expires: 240
User-Agent: Linksys/PAP2T-5.1.6(LS)
Content-Length: 451
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp
v=0
o=- 17789112 17789112 IN IP4 192.168.1.121
s=-
c=IN IP4 192.168.1.121
t=0 0
m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
------------------------------------------------------------------------
send 343 bytes to udp/[192.168.1.121]:5060 at 13:00:32.882684:
------------------------------------------------------------------------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK-50e478ae
From: 1000 <sip:1000 at 192.168.1.120>;tag=2c7a518d12f9370eo0
To: <sip:1020 at 192.168.1.120>
Call-ID: 316156b9-f6b413d2 at 192.168.1.121
CSeq: 102 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
14-49-15 -0500
Content-Length: 0
------------------------------------------------------------------------
2010-06-30 23:00:32.899423 [DEBUG] sofia.c:5975 IP 192.168.1.121
Rejected by acl "domains". Falling back to Digest auth.
2010-06-30 23:00:32.914917 [NOTICE] switch_channel.c:776 New Channel
sofia/internal/1000 at 192.168.1.120
[77a3ab82-8447-11df-93a7-d9ad5b204ca2]
2010-06-30 23:00:32.943135 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.120) Running State Change CS_NEW
2010-06-30 23:00:32.945232 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/1000 at 192.168.1.120) State NEW
2010-06-30 23:00:32.989778 [DEBUG] sofia.c:4293 Channel
sofia/internal/1000 at 192.168.1.120 entering state [received][100]
2010-06-30 23:00:32.997500 [DEBUG] sofia.c:4304 Remote SDP:
v=0
o=- 17789112 17789112 IN IP4 192.168.1.121
s=-
c=IN IP4 192.168.1.121
t=0 0
m=audio 16466 RTP/AVP 0 102 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
2010-06-30 23:00:33.002369 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[G729:18:8000:20]
2010-06-30 23:00:33.005967 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[PCMU:0:8000:20]
2010-06-30 23:00:33.008248 [DEBUG] sofia_glue.c:3877 Audio Codec
Compare [PCMU:0:8000:30]/[GSM:3:8000:20]
2010-06-30 23:00:33.010721 [DEBUG] sofia_glue.c:3924 Substituting
codec PCMU at 30i@8000h
2010-06-30 23:00:33.019002 [DEBUG] sofia_glue.c:2462 Set Codec
sofia/internal/1000 at 192.168.1.120 PCMU/8000 30 ms 240 samples
2010-06-30 23:00:33.038714 [DEBUG] sofia_glue.c:3816 Set 2833 dtmf
send/recv payload to 101
2010-06-30 23:00:33.040992 [DEBUG] sofia.c:4451
(sofia/internal/1000 at 192.168.1.120) State Change CS_NEW -> CS_INIT
2010-06-30 23:00:33.043182 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000 at 192.168.1.120 [BREAK]
2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.120) Running State Change CS_INIT
2010-06-30 23:00:33.045317 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 192.168.1.120) State INIT
2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:83
sofia/internal/1000 at 192.168.1.120 SOFIA INIT
2010-06-30 23:00:33.048752 [DEBUG] mod_sofia.c:117
(sofia/internal/1000 at 192.168.1.120) State Change CS_INIT -> CS_ROUTING
2010-06-30 23:00:33.048752 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000 at 192.168.1.120 [BREAK]
2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/1000 at 192.168.1.120) State INIT going to sleep
2010-06-30 23:00:33.048752 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.120) Running State Change CS_ROUTING
2010-06-30 23:00:33.048752 [DEBUG] switch_channel.c:1474
(sofia/internal/1000 at 192.168.1.120) Callstate Change DOWN -> RINGING
2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000 at 192.168.1.120) State ROUTING
2010-06-30 23:00:33.055456 [DEBUG] switch_channel.c:1333
(sofia/internal/1000 at 192.168.1.120) Callstate Change RINGING -> ACTIVE
2010-06-30 23:00:33.055456 [DEBUG] mod_sofia.c:140
sofia/internal/1000 at 192.168.1.120 SOFIA ROUTING
2010-06-30 23:00:33.055456 [DEBUG] switch_core_state_machine.c:77
sofia/internal/1000 at 192.168.1.120 Standard ROUTING
2010-06-30 23:00:33.055456 [INFO] mod_dialplan_xml.c:331 Processing
1000->1020 in context default
Dialplan: sofia/internal/1000 at 192.168.1.120 parsing
[default->Local_1000_1019] continue=false
Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (FAIL)
[Local_1000_1019] destination_number(1020) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/internal/1000 at 192.168.1.120 parsing
[default->Mobile_1020s] continue=false
Dialplan: sofia/internal/1000 at 192.168.1.120 Regex (PASS)
[Mobile_1020s] destination_number(1020) =~ /^(102[0-9])$/
break=on-false
Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(dialed_extension=1020)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action export(codec_string=GSM)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(codec_string=GSM)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
bridge(user/${dialed_extension}@${domain})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action set_user(1000@${domain})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer()
Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
system(/usr/local/freeswitch/scripts/sms.pl ${smsaccount}
${smspassword} ${smsnumber} 'You have one new voicemail from ${effec$
<action application=)
2010-06-30 23:00:33.110926 [DEBUG] switch_core_state_machine.c:119
(sofia/internal/1000 at 192.168.1.120) State Change CS_ROUTING ->
CS_EXECUTE
2010-06-30 23:00:33.114917 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/1000 at 192.168.1.120 [BREAK]
2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:341
(sofia/internal/1000 at 192.168.1.120) State ROUTING going to sleep
2010-06-30 23:00:33.117257 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/1000 at 192.168.1.120) Running State Change CS_EXECUTE
2010-06-30 23:00:33.128360 [DEBUG] switch_core_state_machine.c:348
(sofia/internal/1000 at 192.168.1.120) State EXECUTE
2010-06-30 23:00:33.129661 [DEBUG] mod_sofia.c:233
sofia/internal/1000 at 192.168.1.120 SOFIA EXECUTE
2010-06-30 23:00:33.134566 [DEBUG] switch_core_state_machine.c:157
sofia/internal/1000 at 192.168.1.120 Standard EXECUTE
EXECUTE sofia/internal/1000 at 192.168.1.120 set(dialed_extension=1020)
2010-06-30 23:00:33.157698 [DEBUG] mod_dptools.c:843
sofia/internal/1000 at 192.168.1.120 SET [dialed_extension]=[1020]
EXECUTE sofia/internal/1000 at 192.168.1.120 bridge(user/1020 at mydns.dyndns.org)
2010-06-30 23:00:33.248458 [DEBUG] switch_ivr_originate.c:1956
variable string 0 = [presence_id=1020 at mydns.dyndns.org]
2010-06-30 23:00:33.260573 [NOTICE] switch_channel.c:776 New Channel
sofia/internal/sip:1020 at 192.168.1.123:5060
[77d7e3b6-8447-11df-93a8-d9ad5b204ca2]
2010-06-30 23:00:33.303487 [DEBUG] mod_sofia.c:3883
(sofia/internal/sip:1020 at 192.168.1.123:5060) State Change CS_NEW ->
CS_INIT
2010-06-30 23:00:33.304564 [DEBUG] switch_core_session.c:1027 Send
signal sofia/internal/sip:1020 at 192.168.1.123:5060 [BREAK]
2010-06-30 23:00:33.327507 [DEBUG] switch_core_state_machine.c:314
(sofia/internal/sip:1020 at 192.168.1.123:5060) Running State Change
CS_INIT
2010-06-30 23:00:33.332010 [DEBUG] switch_core_state_machine.c:338
(sofia/internal/sip:1020 at 192.168.1.123:5060) State INIT
2010-06-30 23:00:33.334621 [DEBUG] mod_sofia.c:83
sofia/internal/sip:1020 at 192.168.1.123:5060 SOFIA INIT
send 1183 bytes to udp/[192.168.1.123]:5060 at 13:00:33.351944:
------------------------------------------------------------------------
INVITE sip:1020 at 192.168.1.123:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.120;rport;branch=z9hG4bKFDpKe5v6j2QcQ
Max-Forwards: 69
From: "1000" <sip:1000 at 192.168.1.120>;tag=Z51Qve55SUHta
To: <sip:1020 at 192.168.1.123:5060>
Call-ID: 4f442091-feea-122d-448a-00e04c0312e9
CSeq: 132833080 INVITE
Contact: <sip:mod_sofia at 192.168.1.120:5060>
User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-9b5778f 2010-06-19
14-49-15 -0500
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info,
sla, include-session-description, presence.winfo, message-summary,
refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 246
X-FS-Support: update_display
Remote-Party-ID: "1000"
<sip:1000 at 192.168.1.120>;party=calling;screen=yes;privacy=off
v=0
o=FreeSWITCH 1277875447 1277875448 IN IP4 192.168.1.120
s=FreeSWITCH
c=IN IP4 192.168.1.120
t=0 0
m=audio 27386 RTP/AVP 0 101 13
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:13 CN/8000
a=ptime:30
It fails after this with Not Acceptable Here / INCOMPATIBLE_DESTINATION
On Mon, Jun 28, 2010 at 5:15 PM, David Ponzone <david.ponzone at gmail.com> wrote:
> Please, retry with a genuine config (the default one would be a wise
> choice).
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
> Service Client IPeva
> tel: 0811 46 26 26
> www.ipeva.fr - www.ipeva-studio.com
> Ce message et toutes les pièces jointes sont confidentiels et établis à
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autorisée est interdite. Tout message électronique est susceptible
> d'altération. IPeva décline toute responsabilité au titre de ce message s'il
> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
> message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>
>
>
> Le 28/06/2010 à 07:37, Mark Campbell-Smith a écrit :
>
> Hi All,
>
> I'm not really sure if I got a firm answer for this one. Is the only
> way to ensure that transcoding is performed is by using
> late-negotiation? Why aren't all my codecs sent in the INVITE message
> to the B-leg (extension 1020)?
>
> Thanks!
>
> On Thu, Jun 24, 2010 at 9:32 AM, Mark Campbell-Smith
> <mcampbellsmith at gmail.com> wrote:
>
> FS version and codecs are shown below, but my config file are probably
>
> quite old. But I guess they should still work?
>
> All codecs are loaded, and the call works if late negotiation is set
>
> on profile internal.
>
> As I wrote above:
>
> The call setup is extension 1000 calls extension 1020
>
> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
>
> FS as the A-leg codec
>
> 2. Extension 1020 only supports GSM codec. The call fails with Not
>
> Acceptable Here.
>
> I forgot to write that Extension 1000 does not support GSM (I want to
>
> force transcoding). Is that why FS is filtering out GSM on the b-leg?
>
> freeswitch at internal> version
>
> FreeSWITCH Version 1.0.head (git-9b5778f 2010-06-19 14-49-15 -0500)
>
> freeswitch at internal> show codecs
>
> type,name,ikey
>
> codec,ADPCM (IMA),mod_voipcodecs
>
> codec,G.711 alaw,CORE_PCM_MODULE
>
> codec,G.711 ulaw,CORE_PCM_MODULE
>
> codec,G.722,mod_voipcodecs
>
> codec,G.723.1 6.3k,mod_g723_1
>
> codec,G.726 16k,mod_voipcodecs
>
> codec,G.726 16k (AAL2),mod_voipcodecs
>
> codec,G.726 24k,mod_voipcodecs
>
> codec,G.726 24k (AAL2),mod_voipcodecs
>
> codec,G.726 32k,mod_voipcodecs
>
> codec,G.726 32k (AAL2),mod_voipcodecs
>
> codec,G.726 40k,mod_voipcodecs
>
> codec,G.726 40k (AAL2),mod_voipcodecs
>
> codec,G.729,mod_com_g729
>
> codec,GSM,mod_voipcodecs
>
> 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_voipcodecs
>
> 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
>
> 25 total.
>
>
> On Wed, Jun 23, 2010 at 11:05 PM, David Ponzone <david.ponzone at gmail.com>
> wrote:
>
> Mark,
>
> I confirm that, as I wrote that wiki page (the early negotiation part) :)
>
> Can you really confirm your FS version ?
>
> The parameter you showed is old.
>
> codec-prefs has been replaced in SIP profiles by:
>
> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>
>
> David Ponzone Direction Technique
>
> email: david.ponzone at ipeva.fr
>
> tel: 01 74 03 18 97
>
> gsm: 06 66 98 76 34
>
> Service Client IPeva
>
> tel: 0811 46 26 26
>
> www.ipeva.fr - www.ipeva-studio.com
>
> Ce message et toutes les pièces jointes sont confidentiels et établis à
>
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>
> non autorisée est interdite. Tout message électronique est susceptible
>
> d'altération. IPeva décline toute responsabilité au titre de ce message s'il
>
> a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce
>
> message, merci de le détruire immédiatement et d'avertir l'expéditeur.
>
>
>
> Le 23/06/2010 à 14:43, Mark Campbell-Smith a écrit :
>
> Check this good wiki page for how FS negotiates codecs (early
>
> negotiation default):
>
> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> I have this set in my internal profile:
>
> <param name="codec-prefs" value="$${global_codec_prefs}"/>
>
> and as stated before vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
>
> Setting late negotiation works (thanks Sergey), but reading the wiki
>
> page, I see the following sentence, which I interpret that GSM should
>
> still be sent:
>
> When FS calls leg B, the list of codecs in outbound-codec-prefs for
>
> the SIP profile is reorganized by pushing the codec negotiated above
>
> for leg A at the top . If B does not accept any of the codecs, the
>
> calls fails, obviously.
>
>
>
> On Wed, Jun 23, 2010 at 10:28 PM, Tony Graziano
>
> <tgraziano at myitdepartment.net> wrote:
>
> I'm a newb to fs, but doesn't codec get neogtiated by the endpoints?
>
> Wouldn't fs only get involved when its media server is referred to?
>
> If the "other endpoint" will only accept G729, doesn't that mean you
>
> need to change that endpoint to also accept G711 or also license G729
>
> in FS?
>
> On 6/23/10, Mark Campbell-Smith <mcampbellsmith at gmail.com> wrote:
>
> Test Setup:
>
> vars.xml:
>
> <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,PCMU,GSM"/>
>
> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,GSM"/>
>
> The call setup is extension 1000 calls extension 1020
>
> 1. Extension 1000 calls with preferred codec PCMU. PCMU is chosen by
>
> FS as the A-leg codec
>
> 2. Extension 1020 only supports GSM codec. The call fails with Not
>
> Acceptable Here.
>
> FS only offers G729 and PCMU to 1020. How do I change the number of
>
> codecs that are offered to an extension? I know I can change the
>
> order in the codec_prefs, but would prefer FS to offer all three
>
> codecs to an extension.
>
> m=audio 23662 RTP/AVP 0 18 101 13
>
> a=rtpmap:0 PCMU/8000
>
> a=rtpmap:18 G729/8000
>
> a=rtpmap:101 telephone-event/8000
>
> a=fmtp:101 0-16
>
> a=rtpmap:13 CN/8000
>
> a=ptime:30
>
> Thanks
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
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>
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>
>
> --
>
> Sent from my mobile device
>
> ======================
>
> Tony Graziano, Manager
>
> Telephone: 434.984.8430
>
> sip: tgraziano at voice.myitdepartment.net
>
> Fax: 434.984.8431
>
> Email: tgraziano at myitdepartment.net
>
> LAN/Telephony/Security and Control Systems Helpdesk:
>
> Telephone: 434.984.8426
>
> sip: helpdesk at voice.myitdepartment.net
>
> Fax: 434.984.8427
>
> Helpdesk Contract Customers:
>
> http://www.myitdepartment.net/gethelp/
>
> Why do mathematicians always confuse Halloween and Christmas?
>
> Because 31 Oct = 25 Dec.
>
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