itf-yamanocuhi / sipml5

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On public domain WebRTC2SIP gateway; browser is not playing RTP #100

Open GoogleCodeExporter opened 9 years ago

GoogleCodeExporter commented 9 years ago
Hi Team,

Browser is not able to play audio when it receives UDP/RTP packets from media 
server via WebRTC2SIP  gateway.
Everything file when all (browser/gateway) are working in local lan. After i 
makes  WebRTC2SIP  to a public ip then browser is lost to play packets.

The only difference i can figure it is in 200OK; The STUN ICE candidate port is 
different from port of RTP/UDP packet received at Browser for the same public 
ip (zzz.zzz.zzz.zzz).

xxx.xxx.xxx.xxx = public ip assigned for gateway at firewall
yyy.yyy.yyy.yyy = private ip assigned for gateway
zzz.zzz.zzz.zzz = STUN ICE ip for private ip yyy.yyy.yyy.yyy

RTP packets source ip is zzz.zzz.zzz.zzz but different source port than the 
200OK.

Is this really matters? Please help me here.
Is it STUN causing is issue? How to enable TURN here ..?

recv=SIP/2.0 200 OK
Via: SIP/2.0/WS 
df7jal23ls0d.invalid;rport;branch=z9hG4bKQZQTfJM2VwWBEdtptwTTiNRXBi8eJGz7
From: "radisys"<sip:8006664567@xxx.xxx.xxx.xxx>;tag=eJ22aiZqnNuclP1M5smU
To: <sip:8003334567@10.201.1.53>;tag=860986086
Contact: 
<sip:8003334567@yyy.yyy.yyy.yyy:5007;transport=ws;ws-src-ip=106.198.109.216;ws-s
rc-port=49320;ws-src-proto=ws>
Call-ID: 2ac99cd9-dcaa-132f-e4be-fb6221a5c125
CSeq: 6356 INVITE
Content-Type: application/sdp
Content-Length: 952
Allow: ACK,BYE,CANCEL,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE

v=0
o=doubango 1983 678901 IN IP4 172.24.30.48
s=-
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 21804 RTP/SAVPF 111 0 8 126
c=IN IP4 yyy.yyy.yyy.yyy
a=ptime:20
a=minptime:20
a=maxptime:20
a=silenceSupp:off - - - -
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; 
sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:126 telephone-event/8000/1
a=fmtp:126 0-16 
a=crypto:0 AES_CM_128_HMAC_SHA1_32 
inline:qTnSuYfh79ZBg3sWKOuPgwF0z3RMaUDDNQXmVZdG
a=sendrecv
a=rtcp-mux
a=ssrc:26023649 cname:2fec35f5487c084099c4cf8ddbd5820e
a=ssrc:26023649 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:26023649 label:doubango@audio
a=ice-ufrag:okDdtE9Ss1ua83m
a=ice-pwd:T7Xv2oiiviuyzixMnmG0g
a=candidate:gEdT88UeL 1 udp 2130706431 yyy.yyy.yyy.yyy 21804 typ host
a=candidate:srflxgEdT 1 udp 1694498815 zzz.zzz.zzz.zzz 13728 typ srflx raddr 
yyy.yyy.yyy.yyy rport 21804
 SIPml-api.js:826
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js:826
setRemoteDescription(answer)
v=0
o=doubango 1983 678901 IN IP4 yyy.yyy.yyy.yyy
s=-
c=IN IP4 yyy.yyy.yyy.yyy
t=0 0
m=audio 21804 RTP/SAVPF 111 0 8 126
c=IN IP4 yyy.yyy.yyy.yyy
a=ptime:20
a=minptime:20
a=maxptime:20
a=silenceSupp:off - - - -
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=48000; sprop-maxcapturerate=48000; stereo=0; 
sprop-stereo=0; useinbandfec=0; usedtx=0
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:126 telephone-event/8000/1
a=fmtp:126 0-16
a=crypto:0 AES_CM_128_HMAC_SHA1_32 
inline:qTnSuYfh79ZBg3sWKOuPgwF0z3RMaUDDNQXmVZdG
a=sendrecv
a=rtcp-mux
a=ssrc:26023649 cname:2fec35f5487c084099c4cf8ddbd5820e
a=ssrc:26023649 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2
a=ssrc:26023649 label:doubango@audio
a=ice-ufrag:okDdtE9Ss1ua83m
a=ice-pwd:T7Xv2oiiviuyzixMnmG0g
a=candidate:gEdT88UeL 1 udp 2130706431 yyy.yyy.yyy.yyy 21804 typ host
a=candidate:srflxgEdT 1 udp 1694498815 zzz.zzz.zzz.zzz 13728 typ srflx raddr 
yyy.yyy.yyy.yyy rport 21804
 SIPml-api.js:826

Original issue reported on code.google.com by skanagav...@gmail.com on 17 Jun 2013 at 8:02