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

[Jonathan Rose (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23145?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=215873#comment-215873 ] 

Jonathan Rose edited comment on ASTERISK-23145 at 2/28/14 3:28 PM:
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I don't believe my patch will fix the problem described as I've been running into it as well and I'm not aware of any fix for it just yet. I currently theorize that it may have something to do with the Controlled/controlling parties of the ICE session described in this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-23026

EDIT:  Actually, I don't seem to be running into this problem anymore at the start of calls from my desk phones to SIPML5 clients.  I'm just getting stuck in SRTP unprotect failure loops when holding/unholding them from SIPML5 when SIPML5 was the called party. So it's worth a shot to test this after all.
                
      was (Author: jrose):
    I don't believe my patch will fix the problem described as I've been running into it as well and I'm not aware of any fix for it just yet. I currently theorize that it may have something to do with the Controlled/controlling parties of the ICE session described in this issue:
https://issues.asterisk.org/jira/browse/ASTERISK-23026

EDIT:  Actually, I don't seem to be running into this problem anymore at the start of calls from my desk phones to SIPML5 clients.  I'm just getting stuck in SRTP unprotect failure loops when holding/unholding them from SIPML5 when SIPML5 was the called party.
                  
> 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, sip_debug_jssip.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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