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TigerYao / sipml5

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Asterisk 11.7.0 + sipml5 + Chrome : one way RTP, no RTP from Chrome #156

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
a) Before posting your issue you MUST answer to the questions otherwise it
will be rejected (invalid status) by us
b) Please check the issue tacker to avoid duplication
c) Please provide network capture (Wireshark) or Javascript console log
if you want quick response

What steps will reproduce the problem?
1.installed asterisk 11.7.0 w/o any error.
2.make calls frome Chrome to SIP phone
3.phone rings and connects but no rtp coming from chrome

What is the expected output? What do you see instead?
expected outcome is audio on both sides.. but rtp flows only one way i.e. from 
sip phone to chrome... no rtp from chrome to sip phone 

What version of the product are you using? On what operating system?
asterisk 11.7.0 on Slackware/Linux 13.37.0
chrome Version 27.0.1453.110 on Slackware/Linux 14.0
and one sip phone ( hard phone) connected to asterisk
and http://sipml5.org/call.htm

Please provide any additional information below.
calls from sip phone to other sip phone works fine with audio on both the 
sides.... But if we make calls from chrome browser to sip phone then rtp flows 
in one direction only... i.e. frome sip phone to chrome and no rtp from chrome 
to sip phone... audio on chrome side works fine and no audio in sip phone....

below is the snapshot of call made from chrome to sip phone

  == WebSocket connection from '203.199.110.89:45401' for protocol 'sip' accepted using version '13'
    -- Registered SIP '704' at 203.199.110.89:45401
       > Saved useragent "IM-client/OMA1.0 sipML5-v1.2014.01.27" for peer 704
[Feb 26 13:04:22] NOTICE[3340]: chan_sip.c:23522 handle_response_peerpoke: Peer 
'704' is now Reachable. (568ms / 2000ms)

   == Using SIP RTP CoS mark 5
    -- Executing [701@local:1] Answer("SIP/704-0000000c", "") in new stack
    -- Executing [701@local:2] Playback("SIP/704-0000000c", "hello-world") in new stack
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038023, ts 000160, 
len 000164)
    -- <SIP/704-0000000c> Playing 'hello-world.gsm' (language 'en')
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038024, ts 000320, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038025, ts 000480, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038026, ts 000640, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038027, ts 000800, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038028, ts 000960, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038029, ts 001120, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038030, ts 001280, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038031, ts 001440, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (type 00, seq 038032, ts 001600, 
len 000164)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038051, ts 
004640, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038052, ts 
004800, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038053, ts 
004960, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038054, ts 
005120, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038055, ts 
005280, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038056, ts 
005440, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038057, ts 
005600, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038058, ts 
005760, len 4294967284)
Sent RTP packet to      203.199.110.89:46739 (via ICE) (type 00, seq 038059, ts 
005920, len 4294967284)
...
...
...
    -- Executing [701@local:3] Dial("SIP/704-00000001", "SIP/701,15,tT") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/701
    -- SIP/701-00000002 is ringing
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056675, ts 
016152, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056676, ts 
016312, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056677, ts 
016472, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056678, ts 
016632, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056679, ts 
016792, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056680, ts 
016952, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056681, ts 
017112, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056682, ts 
017272, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056683, ts 
017432, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056684, ts 
017592, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056685, ts 
017752, len 4294967284)
...
...
...
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056733, ts 
025432, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056734, ts 
025592, len 4294967284)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056735, ts 
025752, len 4294967284)
    -- SIP/701-00000002 answered SIP/704-00000001
       > 0x7f37881903d0 -- Probation passed - setting RTP source address to 203.199.110.73:20500
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005977, ts 9247521, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056736, ts 
9247520, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005978, ts 9247681, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056737, ts 
9247680, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005979, ts 9247841, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056738, ts 
9247840, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005980, ts 9248001, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056739, ts 
9248000, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005981, ts 9248161, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056740, ts 
9248160, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005982, ts 9248321, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056741, ts 
9248320, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005983, ts 9248481, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056742, ts 
9248480, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005984, ts 9248641, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056743, ts 
9248640, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005985, ts 9248801, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056744, ts 
9248800, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005986, ts 9248961, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056745, ts 
9248960, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005987, ts 9249121, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056746, ts 
9249120, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005988, ts 9249281, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056747, ts 
9249280, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005989, ts 9249441, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056748, ts 
9249440, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005990, ts 9249601, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056749, ts 
9249600, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005991, ts 9249761, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056750, ts 
9249760, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005992, ts 9249921, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056751, ts 
9249920, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005993, ts 9250081, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056752, ts 
9250080, len 4294967284)
Got  RTP packet from    203.199.110.73:20500 (type 00, seq 005994, ts 9250241, 
len 000160)
Sent RTP packet to      203.199.110.89:51955 (via ICE) (type 00, seq 056753, ts 
9250240, len 4294967284)

Original issue reported on code.google.com by shad.a.s...@gmail.com on 26 Feb 2014 at 7:46

Attachments:

GoogleCodeExporter commented 9 years ago 
i''d like to add few more comments on this...

 in the above scenario, my asterisk server is an amazon web services(cloud)..
 i tried the same setup on a local virtual machine and everything is working properly, i can see both ways RTP flowing between chrome and sip..

still same setup is not working in AWS...

can someone please help me with this..
thank you in advance.

Original comment by shad.a.s...@gmail.com on 4 Mar 2014 at 9:01

GoogleCodeExporter commented 9 years ago 
I'm running asterisk on EC2 and to make it working on both way audio, I 
manually patched asterisk's src/channels/chan_sip.c file to send the IP address 
of my EC2 machine.

It is in function: add_ice_to_sdp

Original Line: 
ast_str_append(a_buf, 0, "%s ", 
ast_sockaddr_stringify_host(&candidate->address));

New line: 
ast_str_append(a_buf, 0, "%s ", "[ec2-server-ip-address]");

That's the only way I found it to be working.

Original comment by chetan.g...@gmail.com on 22 May 2014 at 11:43

GoogleCodeExporter commented 9 years ago 
Hi,

I'm facing the same situation, then my asterisk server is behind nat (required 
ports are forwarded). My Sipml5 + Chromium 34 has sound, but I can't hear 
anything on my Cisco phone. rtp debug on asterisk shows bidirectional rtp 
stream, but if it try to analize packets with wireshark i can see normal rtp 
packets flowing from asterisk to Chromium, and only UDP STUN request packets 
flowing from Chromium to asterisk. This mens that ICE session is not 
established and don't know how to fix it...

Original comment by virmanta...@lamoda.ru on 18 Jul 2014 at 7:49

GoogleCodeExporter commented 9 years ago 
hi,

anyone found solution? please share

Original comment by 2haf...@gmail.com on 1 Feb 2015 at 5:34