[OpenSIPS-Devel] WebRtc and opensips
Герман Чичикин
1393419 at mail.ru
Fri Aug 21 10:00:48 CEST 2015
Hi, Razvan, thanks for the fast response .
ngep shows:
#
U 192.168.10.62:5060 -> 192.168.10.82:5060
INVITE sip:111111 at 192.168.10.82:5060 SIP/2.0..Record-Route: <sip:192.168.10
.62;r2=on;lr>..Record-Route: <sip:192.168.10.62:8091;transport=ws;r2=on;lr>
..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.7d75f496.0;i=2..Vi
a: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36;branch=z9hG4bK775
6386..Max-Forwards: 68..To: <sip: 111111 at 192.168.10.62 >..From: "222222" <sip
: 222222 at 192.168.10.62 >;tag=pisvn8sql7..Call-ID: 48akg6r345bbtdiofjcg..CSeq:
3313 INVITE..X-Can-Renegotiate: false..Contact: <sip: p06q144b at 192.168.10.3
6:4110;transport=ws;ob>..Content-Type: application/sdp..Session-Expires: 90
..Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER..Supported: tim
er,ice,replaces,outbound..User-Agent: JsSIP 0.7.4..Content-Length: 1096....
v=0..o=mozilla...THIS_IS_SDPARTA-40.0.2 4294967295 0 IN IP4 0.0.0.0..s=-..t
=0 0..a=msid-semantic:WMS *..m=audio 50000 RTP/AVP 109 9 0 8..c=IN IP4 192.
168.10.62..a=end-of-candidates..a=msid:{c2d03933-6c9c-4078-9eeb-b8ac29caea1
6} {38e37d34-8ac9-4b3f-bdaf-228eeb111dd2}..a=rtpmap:109 opus/48000/2..a=rtp
map:9 G722/8000/1..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=ssrc:12013
51425 cname:{ab765227-cdd6-4ce5-aed8-fb7840d18020}..a=sendrecv..a=rtcp:5000
1..a=rtcp-mux..m=video 50020 RTP/AVP 120 126 97..c=IN IP4 192.168.10.62..a=
end-of-candidates..a=fmtp:120 max-fs=12288;max-fr=60..a=fmtp:126 profile-le
vel-id=42e01f;level-asymmetry-allowed=1;packetization-mode=1..a=fmtp:97 pro
file-level-id=42e01f;level-asymmetry-allowed=1.
#
U 192.168.10.82:5060 -> 192.168.10.62:5060
SIP/2.0 100 Trying..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.
7d75f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36
;branch=z9hG4bK7756386..Record-Route: <sip:192.168.10.62;r2=on;lr>..Record-
Route: <sip:192.168.10.62:8091;transport=ws;r2=on;lr>..From: "222222" <sip:
222222 at 192.168.10.62 >;tag=pisvn8sql7..To: <sip: 111111 at 192.168.10.62 >..Call-
ID: 48akg6r345bbtdiofjcg..CSeq: 3313 INVITE..Allow: INVITE, OPTIONS, ACK, B
YE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..Server: SIPPER for PhonerLite..C
ontent-Length: 0....
#
U 192.168.10.82:5060 -> 192.168.10.62:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4
.7d75f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.3
6;branch=z9hG4bK7756386..Record-Route: <sip:192.168.10.62;r2=on;lr>..Record
-Route: <sip:192.168.10.62:8091;transport=ws;r2=on;lr>..From: "222222" <sip
: 222222 at 192.168.10.62 >;tag=pisvn8sql7..To: <sip: 111111 at 192.168.10.62 >;tag=8
062fe20a945e51183b7e539fe35f40c..Call-ID: 48akg6r345bbtdiofjcg..CSeq: 3313
INVITE..Contact: <sip:111111 at 192.168.10.82:5060>..Allow: INVITE, OPTIONS, A
CK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..Server: SIPPER for PhonerLi
te..Content-Length: 0....
#
U 192.168.10.82:5060 -> 192.168.10.62:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bKa9a4.7d75
f496.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36;bra
nch=z9hG4bK7756386..Record-Route: <sip:192.168.10.62;r2=on;lr>..Record-Rout
e: <sip:192.168.10.62:8091;transport=ws;r2=on;lr>..From: "222222" <sip:2222
22 at 192.168.10.62 >;tag=pisvn8sql7..To: <sip: 111111 at 192.168.10.62 >;tag=8062fe
20a945e51183b7e539fe35f40c..Call-ID: 48akg6r345bbtdiofjcg..CSeq: 3313 INVIT
E..Contact: <sip:111111 at 192.168.10.82:5060>..Content-Type: application/sdp.
.Allow: INVITE, OPTIONS, ACK, BYE, CANCEL, INFO, NOTIFY, MESSAGE, UPDATE..S
ession-Expires: 90;refresher=uac..Supported: replaces, timer, from-change..
Server: SIPPER for PhonerLite..Content-Length: 289....v=0..o=- 1976025328
1 IN IP4 192.168.10.82..s=SIPPER for PhonerLite..c=IN IP4 192.168.10.82..t
=0 0..m=audio 5062 RTP/AVP 107 8 0 9..a=rtpmap:107 opus/48000..a=rtpmap:9 G
722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtcp-mux..a=ssrc:19
05944947..a=sendrecv..m=video 0 RTP/AVP 120..
#
U 192.168.10.62:5060 -> 192.168.10.82:5060
ACK sip:111111 at 192.168.10.82:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.62:5
060;branch=z9hG4bKa9a4.7d75f496.2;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid
;received=192.168.10.36;branch=z9hG4bK8369992..Max-Forwards: 68..To: <sip:1
11111 at 192.168.10.62 >;tag=8062fe20a945e51183b7e539fe35f40c..From: "222222" <
sip: 222222 at 192.168.10.62 >;tag=pisvn8sql7..Call-ID: 48akg6r345bbtdiofjcg..CS
eq: 3313 ACK..Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER..Su
pported: outbound..User-Agent: JsSIP 0.7.4..Content-Length: 0....
#
U 192.168.10.62:5060 -> 192.168.10.82:5060
BYE sip:111111 at 192.168.10.82:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.10.62:5
060;branch=z9hG4bK79a4.0ef211d.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;
received=192.168.10.36;branch=z9hG4bK4019217..Max-Forwards: 68..To: <sip:11
1111 at 192.168.10.62 >;tag=8062fe20a945e51183b7e539fe35f40c..From: "222222" <s
ip: 222222 at 192.168.10.62 >;tag=pisvn8sql7..Call-ID: 48akg6r345bbtdiofjcg..CSe
q: 3314 BYE..Reason: SIP ;cause=488; text="Not Acceptable Here"..Allow: INV
ITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER..Supported: outbound..User-
Agent: JsSIP 0.7.4..Content-Length: 0....
#
U 192.168.10.82:5060 -> 192.168.10.62:5060
SIP/2.0 200 OK..Via: SIP/2.0/UDP 192.168.10.62:5060;branch=z9hG4bK79a4.0ef2
11d.0;i=2..Via: SIP/2.0/WS fvnm7l6n6841.invalid;received=192.168.10.36;bran
ch=z9hG4bK4019217..From: "222222" <sip: 222222 at 192.168.10.62 >;tag=pisvn8sql7
..To: <sip: 111111 at 192.168.10.62 >;tag=8062fe20a945e51183b7e539fe35f40c..Call
-ID: 48akg6r345bbtdiofjcg..CSeq: 3314 BYE..Contact: <sip:111111 at 192.168.10.
82:5060>..Server: SIPPER for PhonerLite..Content-Length: 0....
Best regards, Herman
P.S. If I misapply the tool (mailing list)
please sorry and correct me.
>
>
>Message: 1
>Date: Thu, 20 Aug 2015 13:45:59 +0300
>From: ?????? ??????? < 1393419 at mail.ru >
>Subject: [OpenSIPS-Devel] WebRtc and opensips
>To: devel < devel at lists.opensips.org >
>Message-ID: < 1440067559.818370798 at f379.i.mail.ru >
>Content-Type: text/plain; charset="utf-8"
>
> Hi, All!
>
>Having trouble with testing webrtc with a rtpengin. I use opensips.cfg from opensips.org (Documentation -> Tutorials -> WebSocket Transport using OpenSIPS), opensips v2.1, clients - sipml5, jssip demo webphones or my own script in various combinations. Clients register without problems. When a client calls another, the callee joyfully ringings. But when I pick up the phone call is dropped, session ends and no any media. The caller receives a "bye" from opensips.
>Anybody can tell in what direction to dig?
>
>Best regards, Herman.
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>------------------------------
>
>Message: 2
>Date: Thu, 20 Aug 2015 14:30:38 +0300
>From: R?zvan Crainea < razvan at opensips.org >
>Subject: Re: [OpenSIPS-Devel] WebRtc and opensips
>To: devel at lists.opensips.org
>Message-ID: < 55D5BA5E.9030402 at opensips.org >
>Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
>Any chance you could provide a pcap capture of the call?
>
>Best regards,
>
>R?zvan Crainea
>OpenSIPS Solutions
>www.opensips-solutions.com
>
>On 08/20/2015 01:45 PM, ?????? ??????? wrote:
>> Hi, All!
>>
>> Having trouble with testing webrtc with a rtpengin. I use opensips.cfg
>> from opensips.org (Documentation -> Tutorials -> WebSocket Transport
>> using OpenSIPS), opensips v2.1, clients - sipml5, jssip demo webphones
>> or my own script in various combinations. Clients register without
>> problems. When a client calls another, the callee joyfully calls. But
>> when I pick up the phone call is dropped, session ends and no any
>> media. The caller receives a "bye" from opensips.
>> Anybody can tell in what direction to dig?
>>
>> Best regards, Herman.
>>
>>
>> _______________________________________________
>> Devel mailing list
>> Devel at lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/devel
>
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