[OpenSER-Users] problem in audio when using media proxy
Krunal Patel
krunal.patel at ecosmob.com
Fri May 16 10:56:30 CEST 2008
Hi,
Actually I am new to openser.
Here i am sending trace from openser.
So please check out SDP.
Thanking in advance.
Message Body: INVITE
sip:999999999 at voip.domain.com<sip%3A999999999 at voip.domain.com>SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>>
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28079 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 291
v=0
o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35
s=X-Lite
c=IN IP4 122.xxx.xxx.35
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
Message Body: INVITE
sip:999999999 at voip.domain.com<sip%3A999999999 at voip.domain.com>SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>>
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 INVITE
Proxy-Authorization: Digest username="5555555555",realm="voip.domain.com
",nonce="482c8213b23f69ac2b268e593620cb2367dc643b",response="7a7d3cf46a5b7cf90cc623664ca52e4d",uri="
sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1105d
Content-Length: 291
v=0
o=5555555555 3961416264 3961416447 IN IP4 122.xxx.xxx.35
s=X-Lite
c=IN IP4 122.xxx.xxx.35
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
SIP/2.0 183 Session Progress
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061
Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Content-Length: 209
Content-Type: application/sdp
Contact: sip:999999999 at 213.XXX.XXX.28:5060
User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45
00a0ba009b05
v=0
o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28
s=SIP Call
c=IN IP4 213.XXX.XXX.28
t=0 0
m=audio 19462 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 200 OK
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061
Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Content-Length: 209
Content-Type: application/sdp
Supported: replaces
Contact: sip:999999999 at 213.XXX.XXX.28:5060
User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45
00a0ba009b05
v=0
o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28
s=SIP Call
c=IN IP4 213.XXX.XXX.28
t=0 0
m=audio 19462 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
SIP/2.0 200 OK
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED;rport=5061
Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Content-Length: 209
Content-Type: application/sdp
Supported: replaces
Contact: sip:999999999 at 213.XXX.XXX.28:5060
User-Agent: Mediatrix MDD1404 2404 1500 1600 2500 2600 MxSF v3.2.8.45
00a0ba009b05
v=0
o=MxSIP 0 7983 IN IP4 213.XXX.XXX.28
s=SIP Call
c=IN IP4 213.XXX.XXX.28
t=0 0
m=audio 19462 RTP/AVP 0 8 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
Message Body: ACK sip:999999999 at 213.XXX.XXX.28:5060 SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 70
Content-Length: 0
ACK sip:999999999 at 213.XXX.XXX.28:5060 SIP/2.0
Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 69
Content-Length: 0
Message Body: ACK sip:999999999 at 213.XXX.XXX.28:5060 SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 70
Content-Length: 0
ACK sip:999999999 at 213.XXX.XXX.28:5060 SIP/2.0
Record-Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Via: SIP/2.0/UDP 195.XXX.XXX.51;branch=z9hG4bKfacb.bd19b862.2
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport=5061;branch=z9hG4bK384293560A7C9508E28212EC0D72CDFA
From: 5555555555 <sip:5555555555 at voip.domain.com:5061>;tag=2027736222
To: <sip:999999999 at voip.domain.com <sip%3A999999999 at voip.domain.com>
>;tag=5430ebc2c5d6575
Contact: <sip:5555555555 at 122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B at 192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 69
Content-Length: 0
-
Krunal Patel
On Wed, May 14, 2008 at 5:57 PM, Iñaki Baz Castillo <ibc at in.ilimit.es>
wrote:
> El Wednesday 14 May 2008 14:10:46 Krunal Patel escribió:
> > Hi,
> >
> > We are using openser with media proxy and radiator. We're having audio
> > problems when we send calls to mediatrix gateway for termination.
> > Internal calls work fine. We hear audio both side and we can see the
> > session and bytes transfer properly in media proxy monitoring webpage.
> > When we send calls to mediatrix gateway for termination, calls connect
> > properly but we don't hear audio. We see inactive session in mediaproxy
> > status page with no bytes transferring.
> > One doubt I have is it is not entering reply route when it receives
> replies
> > like 183, 180, etc.
> > Has anyone encountered similar issue? How can we fix it?
> > Thanks in advance!
>
> Examine the negoziated SDP (the IP and ports) of each side (caller and
> called)
> and determine if it makes sense or someone is sending audio to a wrong
> place.
>
> --
> Iñaki Baz Castillo
> ibc at in.ilimit.es
>
> _______________________________________________
> Users mailing list
> Users at lists.openser.org
> http://lists.openser.org/cgi-bin/mailman/listinfo/users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.sip-router.org/pipermail/sr-users/attachments/20080516/26ac6713/attachment.htm>
More information about the sr-users
mailing list