Hello, I'm experiencing some strange issue with codecs. I have the following in my vars.xml file: <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/> "inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode. However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters. I'm sending calls like this: <action application="bridge" data="sofia/internal/$1@38.102.93.70"/> Here is a SIP trace of a call: http://pastebin.freeswitch.org/15042 I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291. I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs? my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126 calls flow like this: softphone -> FS -> provider Any help appreciated.
Are those variables being used in the outbound-codec-prefs or codec-prefs params in the sip profile? What does the "sofia status profile NAME" output show for "CODECS OUT"? -Steve On 16 January 2011 20:59, Diego Viola <diego.viola@gmail.com> wrote:
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70 "/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
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From what I can tell the global_codec_prefs variable is used in the internal profile, which is the profile I'm using
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> freeswitch@internal> sofia status profile internal CODECS IN G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM Still I don't know why FS is picking G7221 On Sun, Jan 16, 2011 at 6:45 PM, Steven Ayre <steveayre@gmail.com> wrote:
Are those variables being used in the outbound-codec-prefs or codec-prefs params in the sip profile?
What does the "sofia status profile NAME" output show for "CODECS OUT"?
-Steve
On 16 January 2011 20:59, Diego Viola <diego.viola@gmail.com> wrote:
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
Why not load mod_com_g729 or mod_g729... the only reason an offer would be removed is you DO NOT have the codec loaded. /b On Jan 16, 2011, at 3:57 PM, Diego Viola wrote:
From what I can tell the global_codec_prefs variable is used in the internal profile, which is the profile I'm using
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
freeswitch@internal> sofia status profile internal
CODECS IN G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM
Still I don't know why FS is picking G7221
That's why then. Unless you have disable-transcoding FS will offer all of those to the endpoint. The endpoint picks which one it wants to use. FS will then transcode if necessary. The way to avoid that would be disable-transcoding (FS will pick the aleg codec and then only offer that on the bleg), or use absolute_codec_string. -Steve On 16 January 2011 21:57, Diego Viola <diego.viola@gmail.com> wrote:
From what I can tell the global_codec_prefs variable is used in the internal profile, which is the profile I'm using
<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/>
freeswitch@internal> sofia status profile internal
CODECS IN G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM CODECS OUT G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM
Still I don't know why FS is picking G7221
On Sun, Jan 16, 2011 at 6:45 PM, Steven Ayre <steveayre@gmail.com> wrote:
Are those variables being used in the outbound-codec-prefs or codec-prefs params in the sip profile?
What does the "sofia status profile NAME" output show for "CODECS OUT"?
-Steve
On 16 January 2011 20:59, Diego Viola <diego.viola@gmail.com> wrote:
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h
,G722,PCMU,PCMA,GSM"/>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
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diego, just one thing: remove widband codecs your switch is used for wholesale, not for confrancing that will confuse your vandor thanks Le 16/01/2011 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
-- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313
Done, thanks. freeswitch@internal> sofia status profile internal CODECS IN G729,PCMU,PCMA,GSM CODECS OUT G729,PCMU,PCMA,GSM On Sun, Jan 16, 2011 at 6:55 PM, Meftah Tayeb <tayeb.meftah@gmail.com> wrote:
diego, just one thing: remove widband codecs your switch is used for wholesale, not for confrancing that will confuse your vandor thanks Le 16/01/2011 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
-- Meftah Tayeb inum: +883510001288000 Phone: +13602276297 Fax: +12538020313
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
This can say exactly what ever you wish it to say but if the codec module is NOT loaded it will be removed from the offer. /b On Jan 16, 2011, at 4:15 PM, Diego Viola wrote:
Done, thanks.
freeswitch@internal> sofia status profile internal
CODECS IN G729,PCMU,PCMA,GSM CODECS OUT G729,PCMU,PCMA,GSM
I have mod_com_g729 loaded freeswitch@internal> load mod_com_g729 +OK Reloading XML -ERR [Module already loaded] 2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996 Module mod_com_g729 Already Loaded! freeswitch@internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808 ENUM Reloaded 2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded 530 definitions freeswitch@internal> On Sun, Jan 16, 2011 at 7:18 PM, Brian West <brian@freeswitch.org> wrote:
This can say exactly what ever you wish it to say but if the codec module is NOT loaded it will be removed from the offer.
/b
On Jan 16, 2011, at 4:15 PM, Diego Viola wrote:
Done, thanks.
freeswitch@internal> sofia status profile internal
CODECS IN G729,PCMU,PCMA,GSM CODECS OUT G729,PCMU,PCMA,GSM
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
What does g729_info say? /b On Jan 16, 2011, at 4:22 PM, Diego Viola wrote:
I have mod_com_g729 loaded
freeswitch@internal> load mod_com_g729 +OK Reloading XML -ERR [Module already loaded]
2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996 Module mod_com_g729 Already Loaded! freeswitch@internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808 ENUM Reloaded 2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded 530 definitions
freeswitch@internal>
freeswitch@internal> g729_info Permitted G729 channels: 10 Encoders in use: 0 Decoders in use: 0 freeswitch@internal> On Sun, Jan 16, 2011 at 7:29 PM, Brian West <brian@freeswitch.org> wrote:
What does g729_info say?
/b
On Jan 16, 2011, at 4:22 PM, Diego Viola wrote:
I have mod_com_g729 loaded
freeswitch@internal> load mod_com_g729 +OK Reloading XML -ERR [Module already loaded]
2011-01-16 22:22:49.832772 [WARNING] switch_loadable_module.c:996 Module mod_com_g729 Already Loaded! freeswitch@internal> 2011-01-16 22:22:49.832772 [INFO] mod_enum.c:808 ENUM Reloaded 2011-01-16 22:22:49.832772 [INFO] switch_time.c:954 Timezone reloaded 530 definitions
freeswitch@internal>
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
How do you understand these RTP/AVP numbers? m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 What does the numbers mean? I don't see that on the wiki. Any help appreciated. On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
It's part of the RTP/SDP standards. They're either static or dynamic numbers. Dynamic ones (96-127) are named by name in a=rtpmap lines. Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters -Steve On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70 "/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
_______________________________________________ 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
Oh, and it is allowed in the standard to override a static number with a rtpmap line for a dynamic codec if you run out of numbers in the dynamic range, but the unassigned numbers should be used before any of the assigned ones. -Steve On 16 January 2011 23:21, Steven Ayre <steveayre@gmail.com> wrote:
It's part of the RTP/SDP standards. They're either static or dynamic numbers.
Dynamic ones (96-127) are named by name in a=rtpmap lines.
Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters
-Steve
On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set" data="global_codec_prefs=G729,G7221@32000h,G7221@16000h ,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$ 1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
_______________________________________________ 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
Thanks, I will read that On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre@gmail.com> wrote:
It's part of the RTP/SDP standards. They're either static or dynamic numbers.
Dynamic ones (96-127) are named by name in a=rtpmap lines.
Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters
-Steve
On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
Also, I noticed this on the wiki, in the Early Negotiation example. A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- PCMA/G729/PCMU --------> B Shouldn't that be: A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- G729/PCMA/PCMU --------> B instead? On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola@gmail.com> wrote:
Thanks, I will read that
On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre@gmail.com> wrote:
It's part of the RTP/SDP standards. They're either static or dynamic numbers.
Dynamic ones (96-127) are named by name in a=rtpmap lines.
Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters
-Steve
On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
_______________________________________________ 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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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
http://wiki.freeswitch.org/wiki/Codec_Negotiation On Sun, Jan 16, 2011 at 8:25 PM, Diego Viola <diego.viola@gmail.com> wrote:
Also, I noticed this on the wiki, in the Early Negotiation example.
A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- PCMA/G729/PCMU --------> B
Shouldn't that be:
A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- G729/PCMA/PCMU --------> B
instead?
On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola@gmail.com> wrote:
Thanks, I will read that
On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre@gmail.com> wrote:
It's part of the RTP/SDP standards. They're either static or dynamic numbers.
Dynamic ones (96-127) are named by name in a=rtpmap lines.
Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters
-Steve
On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
_______________________________________________ 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
_______________________________________________ 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
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
Again, read the wiki. After this example, I wrote: -FS checks the proposed codecs in order of priority (as listed in the SDP) against its list of allowed codecs (as configured in inbound-codec-prefs), and selects PCMA as the first authorized codec, so the codecs list becomes: PCMA/G729/PCMU. Basically, that means, that the codec negotiated with A will always be put in the first position of the codecs list sent to B. At least, that's what I observed during my numerous tests at that time. I really recommend you do your own home-work in order to understand it. It really helped me to get it. David Ponzone Direction Technique email: david.ponzone@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 17/01/2011 à 00:25, Diego Viola a écrit :
Also, I noticed this on the wiki, in the Early Negotiation example.
A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- PCMA/G729/PCMU --------> B
Shouldn't that be:
A -------- GSM/PCMA/G729 --------> FS (allowing G729/PCMA/PCMU) -------- G729/PCMA/PCMU --------> B
instead?
On Sun, Jan 16, 2011 at 8:24 PM, Diego Viola <diego.viola@gmail.com> wrote:
Thanks, I will read that
On Sun, Jan 16, 2011 at 8:21 PM, Steven Ayre <steveayre@gmail.com> wrote:
It's part of the RTP/SDP standards. They're either static or dynamic numbers.
Dynamic ones (96-127) are named by name in a=rtpmap lines.
Static numbers are reserved numbers and the a=rtpmap is allowed but optional (although some devices incorrectly require it) The list of reserved static numbers is: http://www.iana.org/assignments/rtp-parameters
-Steve
On 16 January 2011 23:18, Diego Viola <diego.viola@gmail.com> wrote:
How do you understand these RTP/AVP numbers?
m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13
What does the numbers mean? I don't see that on the wiki.
Any help appreciated.
On Sun, Jan 16, 2011 at 7:48 PM, David Ponzone <david.ponzone@ipeva.fr> wrote:
That sounds like a bad reading of the trace. The important line is the 239th: m=audio 23344 RTP/AVP 3 18 98 99 9 0 8 101 13 That means codec 3 is sent first (so GSM), then 18 (so G729). I think you should read a little bit on early negotiation. If leg A has GSM in 1st position, if your inbound codec list allows GSM, and your outbound codec list is : G729, PCM, GSM, than your outbound INVITE to leg B will have GSM in 1st position, because that was requested by leg A. That's in the wiki, it's a part I rewrote myself some weeks ago, in the most readable way I could. David Ponzone Direction Technique email: david.ponzone@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 16/01/2011 à 21:59, Diego Viola a écrit :
Hello,
I'm experiencing some strange issue with codecs. I have the following in my vars.xml file:
<X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G729,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G729,PCMU,PCMA,GSM"/>
"inbound-late-negotiation" and "disable-transcoding" are commented in my internal SIP profile. So I guess I'm in Early Negotiation (default behavior) mode.
However, when I send a call to my provider, and I look at the SIP trace I see that FS is sending another codec, not G729 as I specified in the global_codec_prefs / outbound_codec_prefs parameters.
I'm sending calls like this:
<action application="bridge" data="sofia/internal/$1@38.102.93.70"/>
Here is a SIP trace of a call:
http://pastebin.freeswitch.org/15042
I'm not understanding why FS is sending an INVITE with the G7221 codec in line 240, if I'm telling it explicitly that I want G729 as the priority when possible in the codec prefs options. But I see G729 in the 200 OK in line 291.
I've been told to use absolute_codec_string=G729 in my dialplan or enable late negotiation, but why if I'm already telling it to use G729 in the codec prefs?
my softphone IP: 190.23.80.10 provider IP: 38.102.93.70 FS IP: 77.92.65.126
calls flow like this:
softphone -> FS -> provider
Any help appreciated.
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
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-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
-- Diego Viola Representative of Bridgecom LLC Phone: +595 971 320 520 GTalk: diego.viola@gmail.com MSN: diegoev@msn.com LinkedIn: http://www.linkedin.com/pub/diego-viola/15/886/609 www.bridgecom.com.py
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participants (5)
-
Brian West
-
David Ponzone
-
Diego Viola
-
Meftah Tayeb
-
Steven Ayre