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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-26121?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231064#comment-231064 ] 

Asterisk Team commented on ASTERISK-26121:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> 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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