em 02-20-2019 04:28 PM
Olá amigos,
Peço fortemente a ajuda de todos, estou com problemas em colocar o CP-3905 para funcionar atrás de um firewall pfsense, identifiquei pelo log do pfsense que o aparelho está mandando MTU alto, acima de 1500 e com isso o pfsense está descartando, gostaria de saber qual o parâmetro nas configurações para que eu possa fixar o MTU em 1472
Só detalhando o que ocorre hoje, quando tenho interligação de duas empresas pela internet com um túnel VPN atrás de um firewall pfsense, quando faço uma ligação de uma empresa para outra usando o CP-3905 a pessoa me ouve, mas eu não a ouço, com isso monitorei o tráfego com wireshark e então percebi o MTU acima de 1500 onde o firewall está descartando.
Segue abaixo minha configuração atual
conto com a ajuda de todos e desde já meu agradecimento.
<device>
<deviceProtocol>SIP</deviceProtocol>
<sshUserId>cisco</sshUserId>
<sshPassword>cisco</sshPassword>
<devicePool>
<dateTimeSetting>
<dateTemplate>D/M/YY</dateTemplate>
<timeZone>SA Eastern Standard Time</timeZone>
<ntps>
<ntp>
<!-- SERVIDOR DE DATA e HORA - não altere -->
<name>a.ntp.br</name>
<ntpMode>Unicast</ntpMode>
</ntp>
</ntps>
</dateTimeSetting>
<callManagerGroup>
<members>
<member priority="0">
<callManager>
<ports>
<ethernetPhonePort>2000</ethernetPhonePort>
<sipPort>5060</sipPort>
<securedSipPort>5061</securedSipPort>
</ports>
<!-- IP ou FQDN (host) do SERVIDOR REGISTRO SIP (seu Asterisk, por exemplo) -->
<processNodeName>192.168.0.11</processNodeName>
</callManager>
</member>
</members>
</callManagerGroup>
</devicePool>
<vpnGroup>
<mtu>1472</mtu>
</vpnGroup>
<sipProfile>
<sipProxies>
<backupProxy></backupProxy>
<backupProxyPort>5060</backupProxyPort>
<emergencyProxy></emergencyProxy>
<emergencyProxyPort></emergencyProxyPort>
<outboundProxy></outboundProxy>
<outboundProxyPort></outboundProxyPort>
<registerWithProxy>true</registerWithProxy>
</sipProxies>
<sipCallFeatures>
<cnfJoinEnabled>true</cnfJoinEnabled>
<callForwardURI>x-serviceuri-cfwdall</callForwardURI>
<callPickupURI>x-cisco-serviceuri-pickup</callPickupURI>
<callPickupListURI>x-cisco-serviceuri-opickup</callPickupListURI>
<callPickupGroupURI>x-cisco-serviceuri-gpickup</callPickupGroupURI>
<meetMeServiceURI>x-cisco-serviceuri-meetme</meetMeServiceURI>
<abbreviatedDialURI>x-cisco-serviceuri-abbrdial</abbreviatedDialURI>
<rfc2543Hold>false</rfc2543Hold>
<callHoldRingback>2</callHoldRingback>
<localCfwdEnable>true</localCfwdEnable>
<semiAttendedTransfer>true</semiAttendedTransfer>
<anonymousCallBlock>2</anonymousCallBlock>
<callerIdBlocking>2</callerIdBlocking>
<dndControl>0</dndControl>
<remoteCcEnable>true</remoteCcEnable>
</sipCallFeatures>
<sipStack>
<sipInviteRetx>6</sipInviteRetx>
<sipRetx>10</sipRetx>
<timerInviteExpires>180</timerInviteExpires>
<timerRegisterExpires>120</timerRegisterExpires>
<timerRegisterDelta>5</timerRegisterDelta>
<timerKeepAliveExpires>120</timerKeepAliveExpires>
<timerSubscribeExpires>120</timerSubscribeExpires>
<timerSubscribeDelta>5</timerSubscribeDelta>
<timerT1>500</timerT1>
<timerT2>4000</timerT2>
<maxRedirects>70</maxRedirects>
<remotePartyID>false</remotePartyID>
<userInfo>None</userInfo>
</sipStack>
<autoAnswerTimer>1</autoAnswerTimer>
<autoAnswerAltBehavior>false</autoAnswerAltBehavior>
<autoAnswerOverride>true</autoAnswerOverride>
<transferOnhookEnabled>false</transferOnhookEnabled>
<enableVad>false</enableVad>
<preferredCodec>none</preferredCodec>
<dtmfAvtPayload>101</dtmfAvtPayload>
<dtmfDbLevel>3</dtmfDbLevel>
<dtmfOutofBand>avt</dtmfOutofBand>
<alwaysUsePrimeLine>false</alwaysUsePrimeLine>
<alwaysUsePrimeLineVoiceMail>false</alwaysUsePrimeLineVoiceMail>
<kpml>3</kpml>
<!-- Seu nome com até 13 caracteres, sem espaços -->
<phoneLabel>Secretaria Direito</phoneLabel>
<stutterMsgWaiting>1</stutterMsgWaiting>
<callStats>false</callStats>
<silentPeriodBetweenCallWaitingBursts>10</silentPeriodBetweenCallWaitingBursts>
<disableLocalSpeedDialConfig>false</disableLocalSpeedDialConfig>
<!-- Parâmetros da conta SIP -->
<voipControlPort>5060</voipControlPort>
<startMediaPort>16384</startMediaPort>
<stopMediaPort>32767</stopMediaPort>
<sipLines>
<line button="1">
<featureID>9</featureID>
<!-- IP ou FQDN (host) do SERVIDOR REGISTRO SIP (seu Asterisk, por exemplo) -->
<proxy>192.168.0.11</proxy>
<port>5060</port>
<!-- Usuário SIP ou ramal -->
<featureLabel>3351</featureLabel>
<!-- Usuário SIP ou ramal -->
<name>3351</name>
<!-- Usuário SIP ou ramal -->
<displayName>3351</displayName>
<!-- Usuário SIP ou ramal -->
<authName>3351</authName>
<!-- Usuário SIP ou ramal -->
<contact>3351</contact>
<!-- SENHA da conta SIP -->
<authPassword>SENHA</authPassword>
<autoAnswer>
<autoAnswerEnabled>2</autoAnswerEnabled>
</autoAnswer>
<callWaiting>3</callWaiting>
<sharedLine>false</sharedLine>
<messageWaitingLampPolicy>1</messageWaitingLampPolicy>
<messagesNumber>*97</messagesNumber>
<ringSettingIdle>4</ringSettingIdle>
<ringSettingActive>5</ringSettingActive>
<forwardCallInfoDisplay>
<callerName>true</callerName>
<callerNumber>false</callerNumber>
<redirectedNumber>false</redirectedNumber>
<dialedNumber>true</dialedNumber>
</forwardCallInfoDisplay>
</line>
</sipLines>
<dscpForAudio>184</dscpForAudio>
<ringSettingBusyStationPolicy>0</ringSettingBusyStationPolicy>
<dialTemplate>dialplan.xml</dialTemplate>
<softKeyFile></softKeyFile>
</sipProfile>
<commonProfile>
<phonePassword></phonePassword>
<backgroundImageAccess>true</backgroundImageAccess>
<callLogBlfEnabled>2</callLogBlfEnabled>
</commonProfile>
<!-- Versao do Firmware para auto upgrade (se estiver na mesma pasta TFTP) -->
<loadInformation>CP3905.9-4-1SR3</loadInformation>
<vendorConfig>
<disableSpeaker>false</disableSpeaker>
<disableSpeakerAndHeadset>false</disableSpeakerAndHeadset>
<pcPort>0</pcPort>
<settingsAccess>1</settingsAccess>
<garp>0</garp>
<voiceVlanAccess>0</voiceVlanAccess>
<videoCapability>0</videoCapability>
<autoSelectLineEnable>0</autoSelectLineEnable>
<webAccess>0</webAccess>
<daysDisplayNotActive>1,2,3,4,5,6,7</daysDisplayNotActive>
<displayOnTime>00:00</displayOnTime>
<displayOnDuration>00:00</displayOnDuration>
<displayIdleTimeout>00:00</displayIdleTimeout>
<spanToPCPort>1</spanToPCPort>
<loggingDisplay>1</loggingDisplay>
<loadServer></loadServer>
</vendorConfig>
<userLocale>
<name></name>
<uid></uid>
<langCode>Brazil</langCode>
<version>1.0.0.0-1</version>
<winCharSet>iso-8859-1</winCharSet>
</userLocale>
<networkLocale></networkLocale>
<networkLocaleInfo>
<name>Brazil</name>
<uid></uid>
<version>1.0.0.0-1</version>
</networkLocaleInfo>
<deviceSecurityMode>1</deviceSecurityMode>
<authenticationURL></authenticationURL>
<directoryURL></directoryURL>
<servicesURL></servicesURL>
<idleURL></idleURL>
<informationURL></informationURL>
<messagesURL></messagesURL>
<proxyServerURL></proxyServerURL>
<dscpForSCCPPhoneConfig>96</dscpForSCCPPhoneConfig>
<dscpForSCCPPhoneServices>0</dscpForSCCPPhoneServices>
<dscpForCm2Dvce>96</dscpForCm2Dvce>
<transportLayerProtocol>2</transportLayerProtocol>
<capfAuthMode>0</capfAuthMode>
<capfList>
<capf>
<phonePort>3804</phonePort>
</capf>
</capfList>
<certHash></certHash>
<encrConfig>false</encrConfig>
</device>
em 02-20-2019 06:43 PM
Ola @Luiz Eduardo ,
Eu sei que não é especificamente a resposta para sua preocupação, mas talvez essa resposta ajude :
Your issue isn't latency. Your issue is MTU, because I had an issue with a specific customer and 3905 phones. Especially 3905, are very sensitive to MTU. Unfortunately, it was a long time ago, and I don't recall the exact numbers for adjusting the MSS (with the mss-adjust feature).
I'm 90% sure that you have the issue with the MTU because you said your traffic is flowing through some RF device, and when using RF, in my experience, there MTU is almost always lower then wired network. Especially if you're using some encryption over this RF network, and it's adding extra headers to the packets.
If you don't know what is the maximum MTU that can be achieved from the remote site to your CME in the main site, you can just ping from one site to another while adding to the ping command the following: -l <size>
For example: ping 10.10.10.10 -l 1300 (Windows)
ping 10.10.10.10 -s 1300 (Linux)
And if you are able to ping successfully, just make the size bigger and bigger till you can't. That way you'll know what is your maximum MTU between the 2 sites.
If you have 2 Cisco Routers on both ends, try maybe the following command:
ip tcp adjust-mss 1452
Saudações
em 02-22-2019 08:27 AM
obrigado pela dica, mas realmente não resolve meu problema.
em 04-26-2019 06:32 AM
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