[asterisk-bugs] [JIRA] (ASTERISK-28762) Problem setting up ssl connection. Internal SSL error
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Fri Feb 28 10:00:25 CST 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-28762?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team closed ASTERISK-28762.
------------------------------------
Resolution: Not A Bug
> Problem setting up ssl connection. Internal SSL error
> -----------------------------------------------------
>
> Key: ASTERISK-28762
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-28762
> Project: Asterisk
> Issue Type: Information Request
> Security Level: None
> Components: Core/HTTP
> Affects Versions: 17.2.0
> Environment: CentOS 6 with openssl 1.0.1e
> Asterisk 17.2.0 and WebRTC
> Client browser is Chrome 80.0.3987.87
> Reporter: INVADE International Ltd.
> Severity: Minor
>
> From the forum entry:
> https://community.asterisk.org/t/problem-setting-up-ssl-connection-internal-ssl-error/82633
> Hi. We have a number systems using Asterisk 17 and WebRTC.
> On the systems that use self signed certificates, the following is logged by Asterisk every time a client registers:
> {quote}ERROR[10399] iostream.c: Problem setting up ssl connection: error:00000001:lib(0):func(0):reason(1), Internal SSL error
> ERROR[10399] tcptls.c: Unable to set up ssl connection with peer '192.168.122.1:53700'{quote}
> The errors do not appear to prevent the system from working.
> I’ve enabled debugging and, while there are some debug messages logged after the errors, there is nothing before.
> The errors are not logged on systems that use a certificate from a trusted CA.
> I’ve tested this using both a JsSIP and a sipML5 client, and the results are the same.
> I’m not sure if it helps but, below are the messages from the browser console:
> {quote}s_websocket_server_url= wss://192.168.122.143:8089/ws
> tsk_utils.js?svn=252:116 s_sip_outboundproxy_url=(null)
> tsk_utils.js?svn=252:116 b_rtcweb_breaker_enabled=yes
> tsk_utils.js?svn=252:116 b_click2call_enabled=no
> tsk_utils.js?svn=252:116 b_early_ims=yes
> tsk_utils.js?svn=252:116 b_enable_media_stream_cache=yes
> tsk_utils.js?svn=252:116 o_bandwidth={}
> tsk_utils.js?svn=252:116 o_video_size={}
> tsk_utils.js?svn=252:116 SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:192.168.122.143>', impi='100', impu='"Simon"<sip:100 at 192.168.122.143>'
> tsk_utils.js?svn=252:116 Connecting to ' wss://192.168.122.143:8089/ws'
> tsk_utils.js?svn=252:116 ==stack event = starting
> tsk_utils.js?svn=252:116 __tsip_transport_ws_onopen
> tsk_utils.js?svn=252:116 ==stack event = started
> tsk_utils.js?svn=252:116 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
> tsk_utils.js?svn=252:116 SEND: REGISTER sip:192.168.122.143 SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyFqsoLUUFpedAcVEyZN4cBtFTu9Gt9G9;rport
> From: "Simon"<sip:100 at 192.168.122.143>;tag=kMuhaZy6Q0Wvw40VKeLZ
> To: "Simon"<sip:100 at 192.168.122.143>
> Contact: "Simon"<sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
> Call-ID: ffac53bd-a6b2-9176-63b1-ff8ed7c5fa26
> CSeq: 28995 REGISTER
> Content-Length: 0
> Max-Forwards: 70
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
> Supported: path
> tsk_utils.js?svn=252:116 __tsip_transport_ws_onmessage
> tsk_utils.js?svn=252:116 recv=SIP/2.0 401 Unauthorized
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53702;received=192.168.122.1;branch=z9hG4bKyFqsoLUUFpedAcVEyZN4cBtFTu9Gt9G9
> From: "Simon"<sip:100 at 192.168.122.143>;tag=kMuhaZy6Q0Wvw40VKeLZ
> To: "Simon"<sip:100 at 192.168.122.143>;tag=z9hG4bKyFqsoLUUFpedAcVEyZN4cBtFTu9Gt9G9
> Call-ID: ffac53bd-a6b2-9176-63b1-ff8ed7c5fa26
> CSeq: 28995 REGISTER
> Content-Length: 0
> WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1581088246/deff5cad6eaa2e080cc3a29e7e48d1eb",opaque="1e37dbb763d85954",stale=FALSE,algorithm=md5
> Server: Asterisk PBX 17.2.0
> tsk_utils.js?svn=252:116 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
> tsk_utils.js?svn=252:116 SEND: REGISTER sip:192.168.122.143 SIP/2.0
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKaQerbmtNuKAHJsnBlJlOUbWnwsgSb0b0;rport
> From: "Simon"<sip:100 at 192.168.122.143>;tag=kMuhaZy6Q0Wvw40VKeLZ
> To: "Simon"<sip:100 at 192.168.122.143>
> Contact: "Simon"<sips:100 at df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
> Call-ID: ffac53bd-a6b2-9176-63b1-ff8ed7c5fa26
> CSeq: 28996 REGISTER
> Content-Length: 0
> Max-Forwards: 70
> Authorization: Digest username="100",realm="asterisk",nonce="1581088246/deff5cad6eaa2e080cc3a29e7e48d1eb",uri="sip:192.168.122.143",response="33ab4ea03ce85d9b4ac09058e36992e9",algorithm=md5,cnonce="eb2396623ba609a63f39f28a3413ab43",opaque="1e37dbb763d85954",qop=auth,nc=00000001
> User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
> Organization: Doubango Telecom
> Supported: path
> tsk_utils.js?svn=252:116 ==session event = connecting
> 2tsk_utils.js?svn=252:116 ==session event = sent_request
> tsk_utils.js?svn=252:116 __tsip_transport_ws_onmessage
> tsk_utils.js?svn=252:116 recv=SIP/2.0 200 OK
> Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53702;received=192.168.122.1;branch=z9hG4bKaQerbmtNuKAHJsnBlJlOUbWnwsgSb0b0
> From: "Simon"<sip:100 at 192.168.122.143>;tag=kMuhaZy6Q0Wvw40VKeLZ
> To: "Simon"<sip:100 at 192.168.122.143>;tag=z9hG4bKaQerbmtNuKAHJsnBlJlOUbWnwsgSb0b0
> Contact: <sips:100 at 192.168.122.1:53702;transport=ws;rtcweb-breaker=yes>;expires=199
> Call-ID: ffac53bd-a6b2-9176-63b1-ff8ed7c5fa26
> CSeq: 28996 REGISTER
> Content-Length: 0
> Date: 07 Feb 2020 15:10:46 GMT;07
> Server: Asterisk PBX 17.2.0
> tsk_utils.js?svn=252:116 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
> tsk_utils.js?svn=252:116 ==session event = connected
> tsk_utils.js?svn=252:116 __tsip_transport_ws_onmessage{quote}
> The server is running CentOS 6 with openssl 1.0.1e, and the client browser is Chrome 80.0.3987.87.
> I’m trying to determine:
> What is the cause of the errors.
> Are they actually causing any problems.
> If anyone is able to answer either of these messages it is much appreciated.
> I can provide additional info if required.
> Thanks in advance.
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