[asterisk-bugs] [JIRA] (ASTERISK-26121) No voice for voice calling using Android portsip

[Joshua Colp (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26121?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231066#comment-231066 ] 

Joshua Colp commented on ASTERISK-26121:
----------------------------------------

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information



> No voice for voice calling using Android portsip
> ------------------------------------------------
>
>                 Key: ASTERISK-26121
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26121
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 11.21.2
>         Environment: Software platform
>            Reporter: Dhananjay Arun HArel
>
> Cant get the voice while calling from Android Apps with PORTSIP client sdk
> Following are the Asterisk server logs:
> Called SIP/50000066
>    -- SIP/50000066-00000001 is ringing
>    -- SIP/50000066-00000001 is ringing
>    -- Registered SIP '40000029' at 115.254.41.117:10291
>       > Saved useragent "PortSIP SDK for IOS" for peer 40000029
> [2016-06-17 14:13:52] NOTICE[16459]: chan_sip.c:23767 handle_response_peerpoke: Peer '40000029' is now Reachable. (605ms / 2000ms)
> *CLI> 
> *CLI>        > 0x7fe3d4017a00 -- Probation passed - setting RTP source address to 115.248.199.61:20302
>       > 0x7fe3d4017a00 -- Probation passed - setting RTP source address to 115.248.199.61:20302
>       > 0x7fe3d401f3e0 -- Probation passed - setting RTP source address to 115.248.199.61:43562
> [2016-06-17 14:13:55] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:43562'
>    -- SIP/50000066-00000001 answered SIP/50000068-00000000
>       > 0x7fe3d4017a00 -- Probation passed - setting RTP source address to 115.248.199.61:20302
> [2016-06-17 14:13:55] WARNING[16468][C-00000000]: abstract_jb.c:428 create_jb: Failed to put first frame in the jitterbuffer on channel 'SIP/50000068-00000000'
>    -- adaptive jitterbuffer created on channel SIP/50000068-00000000
> [2016-06-17 14:13:55] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:20302'
> *CLI> 
> *CLI> 
> *CLI> 
> *CLI>     -- Started music on hold, class 'default', on SIP/50000068-00000000
>       > 0x7fe3d401f3e0 -- Probation passed - setting RTP source address to 115.248.199.61:43562
> [2016-06-17 14:13:55] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:43562'
>       > 0x7fe3f8087960 -- Probation passed - setting RTP source address to 115.248.199.61:20848
> [2016-06-17 14:13:55] WARNING[16468][C-00000000]: abstract_jb.c:428 create_jb: Failed to put first frame in the jitterbuffer on channel 'SIP/50000066-00000001'
>    -- adaptive jitterbuffer created on channel SIP/50000066-00000001
> [2016-06-17 14:13:55] WARNING[16468][C-00000000]: chan_iax2.c:1184 jb_warning_output: Resyncing the jb. last_delay 0, this delay -280200120, threshold 1000, new offset 280200120
>       > 0x7fe3f80066d0 -- Probation passed - setting RTP source address to 115.248.199.61:43930
> [2016-06-17 14:13:56] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:43930'
>    -- Started music on hold, class 'default', on SIP/50000066-00000001
>       > 0x7fe3f8087960 -- Probation passed - setting RTP source address to 115.248.199.61:20848
> [2016-06-17 14:13:56] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:20848'
>       > 0x7fe3f80066d0 -- Probation passed - setting RTP source address to 115.248.199.61:43930
> [2016-06-17 14:13:56] NOTICE[16468][C-00000000]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from '115.248.199.61:43930'



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