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

[Rusty Newton (JIRA)]

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

Rusty Newton updated ASTERISK-23145:
------------------------------------

    Description: 
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

  was:
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_config.png

    
> 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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