[asterisk-bugs] [JIRA] (ASTERISK-24143) pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24143?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=221560#comment-221560 ] 

Matt Jordan edited comment on ASTERISK-24143 at 8/14/14 9:53 AM:
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{noformat}
[101]
type=endpoint
disallow=all
allow=ulaw
context=default
auth=101
aors=101
media_encryption=dtls
dtls_verify=fingerprint
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes

[101]
type=aor
max_contacts=2
remove_existing=yes

[101]
type=auth
auth_type=userpass
password=101
username=101
{noformat}

I am using encryption as voice does not go if encryption is not enabled . But the problem comes only when the call is disconnected from the callee end . If caller disconnects then both legs close normally .


was (Author: agupta):
[101]
type=endpoint
disallow=all
allow=ulaw
context=default
auth=101
aors=101
media_encryption=dtls
dtls_verify=fingerprint
dtls_cert_file=/etc/asterisk/keys/asterisk.pem
dtls_ca_file=/etc/asterisk/keys/ca.crt
dtls_setup=actpass
use_avpf=yes
ice_support=yes
media_use_received_transport=yes

[101]
type=aor
max_contacts=2
remove_existing=yes

[101]
type=auth
auth_type=userpass
password=101
username=101

I am using encryption as voice does not go if encryption is not enabled . But the problem comes only when the call is disconnected from the callee end . If caller disconnects then both legs close normally .

> pjsip: Outbound call to WebRTC UA fails to transmit ACK on received 200 OK
> --------------------------------------------------------------------------
>
>                 Key: ASTERISK-24143
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24143
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip, Resources/res_pjsip_transport_websocket
>    Affects Versions: 12.4.0
>         Environment: Ubuntu 14.04 
> PJProject(https://github.com/asterisk/pjproject 06/JUN/14)
> --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr --with-external-srtp CFLAGS="-g -DNDEBUG" 
> chromium 35.0.1916.153(rev274914) (launch options: --use-fake-ui-for-media-stream --disable-webrtc-encryption)
> SIPml-api.js?svn=224
>            Reporter: Aleksei Kulakov
>            Assignee: Rusty Newton
>         Attachments: 354ChromeConsole.log, debuglog.1, pjsip.conf
>
>
> 1. WebRTC callee accepts call from caller of any type.
> 2. Caller hangup.
> 3. WebRTC callee does not get BYE sip packet(staying "in call" forever) and asterisk dumps following message to log:
> {quote}
> WARNING[348] pjsip: tsx0xb68399d4 ...Failed to send Request msg BYE/cseq=9511 (tdta0xb6804798)! err=320053 (DNS "Name Error" (PJLIB_UTIL_EDNS_NXDOMAIN))
> {quote}



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