search

innovatormrityunjay / webrtc2sip

Automatically exported from code.google.com/p/webrtc2sip
0 stars 0 forks source link

Sound decode problem #44

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago 
What steps will reproduce the problem?

1. Call started by caller <SipML5 client>
2. Callee answers the call <SIP Server>
3. Media packages start flowing via webrtc2sip

What is the expected output? What do you see instead?

Expected scenario;

  - SIP Server received a call
  - SIP Server answers the call and starts transmitting pre-recorded audio track with G.711 A-Law (pcma) codec.
  - Caller listens the callee's audio message

Instead of expected scenario, during audio transmission from SIP server 
received sound played like cluttered (as one of the webrtc2sip user Anton said, 
i couldn't come up with better word :) ). I might phrase cluttered as a potato 
robot with low on batteries from Portal game.

What version of the product are you using? On what operating system?

Product stack;

  - SIP Server with G.711 A-Law (pcma) codec support
  - webrtc2sip v2.0 (Running on Ubuntu 12.04 LTS)
  - SipML5 live demo svn.20 (Running on Chrome)

Please provide any additional information below.

my webrtc2sip configuration;

  <debug-level>ERROR</debug-level>

  <transport>udp;*;10060</transport>
  <transport>ws;*;10060</transport>
  <transport>wss;*;10062</transport>

  <enable-100rel>no</enable-100rel>
  <enable-media-coder>no</enable-media-coder>
  <enable-videojb>yes</enable-videojb>
  <rtp-buffsize>65535</rtp-buffsize>
  <avpf-tail-length>100;400</avpf-tail-length>
  <srtp-mode>optional</srtp-mode>

  <codecs>pcma;pcmu;gsm;vp8;h264-bp;h264-mp;h263;h263+</codecs>

my sipML5 configuration;

  - Disable Video -> checked
  - Enable RTCWeb Breaker -> checked
  - WebSocket Server URL -> 'ws://example.com:10060'
  - SIP outbound Proxy URL -> ''

Original issue reported on code.google.com by gokhanba...@gmail.com on 7 Jan 2013 at 11:14

GoogleCodeExporter commented 9 years ago 
It turns out RTP ptime variable is differennt from server.

mine was;

  - a=ptime:20

server's was:

  - a=ptime:40

there was 20 byte of extra package data unreadable for sip client.
Although i searched i couldn't come up with a solution to change ptime 
variable. How can i modify ptime?

Thanks in advance,

Original comment by gokhanba...@gmail.com on 8 Jan 2013 at 9:35

GoogleCodeExporter commented 9 years ago 
[deleted comment]
GoogleCodeExporter commented 9 years ago

Original comment by boss...@yahoo.fr on 9 Jan 2013 at 10:44