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

[Dhananjay Arun HArel (JIRA)]

Dhananjay Arun HArel created ASTERISK-26121:
-----------------------------------------------

             Summary: 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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