14 May
2008
14 May
'08
8:10 a.m.
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!
--
regards,
Krunal Patel
14 May
14 May
8:27 a.m.
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(a)in.ilimit.es
16 May
16 May
4:56 a.m.
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@voip.domain.com<sip%3A999999999@voip.domain.com>SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK7F77064E8D8FD3FD41779B0F67EF5C05
From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>>
Contact: <sip:5555555555@122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)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@voip.domain.com<sip%3A999999999@voip.domain.com>SIP/2.0
Via: SIP/2.0/UDP
122.xxx.xxx.35:5061;rport;branch=z9hG4bK200ED8B65B4AFC558C060A0A8B5FE1ED
From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>>
Contact: <sip:5555555555@122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10
CSeq: 28080 INVITE
Proxy-Authorization: Digest username="5555555555",realm="voip.domain.com
",nonce="482c8213b23f69ac2b268e593620cb2367dc643b",response="7a7d3cf46a5b7cf90cc623664ca52e4d",uri="
sip:999999999@voip.domain.com <sip%3A999999999(a)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(a)192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)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@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(a)192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)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@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(a)192.168.1.10
CSeq: 28080 INVITE
From: 5555555555 <sip:5555555555@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)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@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@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@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>
;tag=5430ebc2c5d6575
Contact:
<sip:5555555555@122.xxx.xxx.35:5061>
Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 70
Content-Length: 0
ACK sip:999999999@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@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>
;tag=5430ebc2c5d6575
Contact:
<sip:5555555555@122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 69
Content-Length: 0
Message Body: ACK sip:999999999@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@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>
;tag=5430ebc2c5d6575
Contact:
<sip:5555555555@122.xxx.xxx.35:5061>
Route: <sip:voip.domain.com:5060;nat=yes;ftag=2027736222;lr=on>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)192.168.1.10
CSeq: 28080 ACK
Max-Forwards: 70
Content-Length: 0
ACK sip:999999999@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@voip.domain.com:5061>;tag=2027736222
To: <sip:999999999@voip.domain.com <sip%3A999999999(a)voip.domain.com>
;tag=5430ebc2c5d6575
Contact:
<sip:5555555555@122.xxx.xxx.35:5061>
Call-ID: 64E8EEAF-6014-1CCB-98A5-1185295C388B(a)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(a)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(a)in.ilimit.es
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
19 May
19 May
5:44 p.m.
On May 14, 2008, at 2:10 PM, Krunal Patel wrote:
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?
Actually it's pretty simple, you need to handle the 18x messages just
like a 200 OK message!
--
Andreas Sikkema
20 May
20 May
12:37 a.m.
Thanks a lot Andreas
Now it is fixed
On Tue, May 20, 2008 at 3:14 AM, Andreas Sikkema <h323(a)ramdyne.nl> wrote:
On May 14, 2008, at 2:10 PM, Krunal Patel wrote:
--
Thanks & regards,
Krunal Patel
Software Engineer
Ecosmob Technologies Pvt. Ltd.
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?
Actually it's pretty simple, you need to handle the 18x messages just
like a 200 OK message!
--
Andreas Sikkema
_______________________________________________
Users mailing list
Users(a)lists.openser.org
http://lists.openser.org/cgi-bin/mailman/listinfo/users
5931
days inactive
5937
days old
4 comments
3 participants
participants (3)
-
Andreas Sikkema -
Iñaki Baz Castillo -
Krunal Patel