[asterisk-bugs] [JIRA] (ASTERISK-23145) Sporadic one way audio between a SIP hard phone and a SIPML5 browser client on LAN

[Andrew Nagy (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-23145?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Andrew Nagy updated ASTERISK-23145:
-----------------------------------

    Attachment: sip_debug_jssip.txt

SIP Debug on outbound call while using jssip in the browser
                
> Sporadic one way audio between a SIP hard phone and a SIPML5 browser client on LAN
> ----------------------------------------------------------------------------------
>
>                 Key: ASTERISK-23145
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-23145
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General, Channels/chan_sip/WebSocket, Resources/res_rtp_asterisk
>    Affects Versions: SVN, 12.0.0
>         Environment: Asterisk SVN-branch-12-r405587
> Chrome Browser Version 32.0.1700.77
> SIPML5 demo http://sipml5.org/call.htm?svn=203
>            Reporter: Rusty Newton
>            Severity: Critical
>         Attachments: extensions.txt, full.txt, http.txt, messages.txt, rtp_fail.pcap, rtp.txt, sipml5_config1.png, sipml5_config2.png, sip.txt
>
>
> Ran across one way audio issues that I believe stem from a bug while testing calls between two SIP clients (6001 and sipml5_chrome in sip.conf) in the process of building some documentation.
> h3. Endpoints in sip.conf
> *6001* is a Digium D40
> *sipml5_chrome* is SIPML5 on the sipml5.org demo running on Chrome.
> h3. Problem description
>  * When *calling from 6001 to sipml5_chrome*, approximately one out of every three to six calls has one way audio, with audio flowing from 6001 to sipml5_chrome, but not the other way around.
>  * When *calling from sipml5_chrome to 6001*, I appeared to get two way audio every time.
> h3. Environment
> SIPML5 is running on a Chrome browser, on the machine running Asterisk.
> The Digium phone is on the local LAN. There is no hardware or software firewalls or NAT going on.
> h3. Files attached
> h5. Config files
> extensions.txt
> http.txt
> rtp.txt
> sip.txt
> h5. Asterisk logs
> full.txt
> messages.txt
> h5. Packet capture
> rtp_fail.pcap
> The pcap contains several calls. Out of the six calls at the end, the first five had one-way audio, and *the very last call had two-way audio* (should be the very last call in the pcap.
> h5. SIPML5 configuration
> sipml5_config1.png
> sipml5_config2.png

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