[sip-comm-dev] Sip Calls to PSTN early Rtp media SDP 183


#1

Hi

I'm testing several Sip Server and i noticed that for the call progress is enable only the 180SDP method that reproduce local ringback for alerting the call progress. SIP calls to PSTN numbers should be able to accept the 183SDP signal to reproduce remote early tone to alert call progress. So when call to PSTN, if the remote party respond with 183SDP you should play the remote early media and not the local ringback.

Thanks
Fabio Galdi

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#2

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the call
progress is enable only the 180SDP method that reproduce local
ringback for alerting the call progress. SIP calls to PSTN numbers
should be able to accept the 183SDP signal to reproduce remote early
tone to alert call progress. So when call to PSTN, if the remote
party respond with 183SDP you should play the remote early media and
not the local ringback.

SIP Communicator already supports 183 Session Progress responses. If you
are having problems with this then you might want to send us a wireshark
trace containing a session that includes early media so that we could
try and see where the problem comes from.

Cheers,
Emil

···

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#3

Hi Emil,

thanks very much for the answer, in attach the cap file i made from my PC, i see in the capture that there are both 183 and 183 SDP signals, i'm not sure at this point if the problem comes from the SIP Server, but other clients always reproduce the correct session progress so this is the first time i experience this problem.
Hope you can find out the reason :slight_smile:

Many Thanks
Fabio

session-progress.pcap (116 KB)


#4

На 21.10.10 14:44, Fabio Telme написа:

Hi Emil,

thanks very much for the answer, in attach the cap file i made from
my PC, i see in the capture that there are both 183 and 183 SDP
signals, i'm not sure at this point if the problem comes from the SIP
Server, but other clients always reproduce the correct session
progress so this is the first time i experience this problem. Hope
you can find out the reason :slight_smile:

You server doesn't seem to be sending any RTP traffic before the 200 OK
so there's simply no early media for SC to play.

You'd have to check the server configuration and logs to understand why
this is happening. You can also try and see whether there are any
differences in the dump when doing the same thing with clients that
receive the early media in your case.

Hope this helps,
Emil

Many Thanks Fabio

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the call
progress is enable only the 180SDP method that reproduce local
ringback for alerting the call progress. SIP calls to PSTN
numbers should be able to accept the 183SDP signal to reproduce
remote early tone to alert call progress. So when call to PSTN,
if the remote party respond with 183SDP you should play the
remote early media and not the local ringback.

SIP Communicator already supports 183 Session Progress responses.
If you are having problems with this then you might want to send us
a wireshark trace containing a session that includes early media so
that we could try and see where the problem comes from.

Cheers, Emil

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···

Il giorno 21/ott/2010, alle ore 12.55, Emil Ivov ha scritto:
For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#5

Hi Emil,

sorry for my earlier email, but i did not noticed that SC plays both ring, the local and remote at the same time, Marc how is it for You ?

Thanks
Fabio

···

Il giorno 21/ott/2010, alle ore 23.23, Emil Ivov ha scritto:

Hey Fabio,

In case you hadn't noticed, this is about the resolution of the "early
media" problem that you reported earlier, and that we have now fixed:

https://sip-communicator.dev.java.net/servlets/ReadMsg?list=dev&msgNo=9699

Cheers,
Emil

На 21.10.10 14:55, Emil Ivov написа:

На 21.10.10 14:44, Fabio Telme написа:

Hi Emil,

thanks very much for the answer, in attach the cap file i made from
my PC, i see in the capture that there are both 183 and 183 SDP
signals, i'm not sure at this point if the problem comes from the SIP
Server, but other clients always reproduce the correct session
progress so this is the first time i experience this problem. Hope
you can find out the reason :slight_smile:

You server doesn't seem to be sending any RTP traffic before the 200 OK
so there's simply no early media for SC to play.

You'd have to check the server configuration and logs to understand why
this is happening. You can also try and see whether there are any
differences in the dump when doing the same thing with clients that
receive the early media in your case.

Hope this helps,
Emil

Many Thanks Fabio

Il giorno 21/ott/2010, alle ore 12.55, Emil Ivov ha scritto:

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the call
progress is enable only the 180SDP method that reproduce local
ringback for alerting the call progress. SIP calls to PSTN
numbers should be able to accept the 183SDP signal to reproduce
remote early tone to alert call progress. So when call to PSTN,
if the remote party respond with 183SDP you should play the
remote early media and not the local ringback.

SIP Communicator already supports 183 Session Progress responses.
If you are having problems with this then you might want to send us
a wireshark trace containing a session that includes early media so
that we could try and see where the problem comes from.

Cheers, Emil

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dev-help@sip-communicator.dev.java.net

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For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#6

Hey Fabio,

На 21.10.10 23:58, Fabio Telme написа:

Hi Emil,

sorry for my earlier email, but i did not noticed that SC plays both
ring,

Should be alright now.

Cheers,
Emil

the local and remote at the same time, Marc how is it for You
?

Thanks Fabio

Hey Fabio,

In case you hadn't noticed, this is about the resolution of the
"early media" problem that you reported earlier, and that we have
now fixed:

https://sip-communicator.dev.java.net/servlets/ReadMsg?list=dev&msgNo=9699

Cheers,

Emil

На 21.10.10 14:55, Emil Ivov написа:

На 21.10.10 14:44, Fabio Telme написа:

Hi Emil,

thanks very much for the answer, in attach the cap file i made
from my PC, i see in the capture that there are both 183 and
183 SDP signals, i'm not sure at this point if the problem
comes from the SIP Server, but other clients always reproduce
the correct session progress so this is the first time i
experience this problem. Hope you can find out the reason :slight_smile:

You server doesn't seem to be sending any RTP traffic before the
200 OK so there's simply no early media for SC to play.

You'd have to check the server configuration and logs to
understand why this is happening. You can also try and see
whether there are any differences in the dump when doing the same
thing with clients that receive the early media in your case.

Hope this helps, Emil

Many Thanks Fabio

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the
call progress is enable only the 180SDP method that
reproduce local ringback for alerting the call progress.
SIP calls to PSTN numbers should be able to accept the
183SDP signal to reproduce remote early tone to alert call
progress. So when call to PSTN, if the remote party respond
with 183SDP you should play the remote early media and not
the local ringback.

SIP Communicator already supports 183 Session Progress
responses. If you are having problems with this then you
might want to send us a wireshark trace containing a session
that includes early media so that we could try and see where
the problem comes from.

Cheers, Emil

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dev-help@sip-communicator.dev.java.net

---------------------------------------------------------------------

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For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

···

Il giorno 21/ott/2010, alle ore 23.23, Emil Ivov ha scritto:

Il giorno 21/ott/2010, alle ore 12.55, Emil Ivov ha scritto:

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#7

На 22.10.10 09:42, Fabio Telme написа:

Hi Emil

i've just installed the latest build but the double ringtone still happen
hope it can be fixed :slight_smile:

Well it appears that the server is sending the 180 after the 183 and not
the other way around.

Should be OK in build 3037.

Cheers,
Emil

···

Thanks
Fabio
Il giorno 22/ott/2010, alle ore 01.47, Emil Ivov ha scritto:

Hey Fabio,

На 21.10.10 23:58, Fabio Telme написа:

Hi Emil,

sorry for my earlier email, but i did not noticed that SC plays both
ring,

Should be alright now.

Cheers,
Emil

the local and remote at the same time, Marc how is it for You
?

Thanks Fabio

Il giorno 21/ott/2010, alle ore 23.23, Emil Ivov ha scritto:

Hey Fabio,

In case you hadn't noticed, this is about the resolution of the
"early media" problem that you reported earlier, and that we have
now fixed:

https://sip-communicator.dev.java.net/servlets/ReadMsg?list=dev&msgNo=9699

Cheers,

Emil

На 21.10.10 14:55, Emil Ivov написа:

На 21.10.10 14:44, Fabio Telme написа:

Hi Emil,

thanks very much for the answer, in attach the cap file i made
from my PC, i see in the capture that there are both 183 and
183 SDP signals, i'm not sure at this point if the problem
comes from the SIP Server, but other clients always reproduce
the correct session progress so this is the first time i
experience this problem. Hope you can find out the reason :slight_smile:

You server doesn't seem to be sending any RTP traffic before the
200 OK so there's simply no early media for SC to play.

You'd have to check the server configuration and logs to
understand why this is happening. You can also try and see
whether there are any differences in the dump when doing the same
thing with clients that receive the early media in your case.

Hope this helps, Emil

Many Thanks Fabio

Il giorno 21/ott/2010, alle ore 12.55, Emil Ivov ha scritto:

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the
call progress is enable only the 180SDP method that
reproduce local ringback for alerting the call progress.
SIP calls to PSTN numbers should be able to accept the
183SDP signal to reproduce remote early tone to alert call
progress. So when call to PSTN, if the remote party respond
with 183SDP you should play the remote early media and not
the local ringback.

SIP Communicator already supports 183 Session Progress
responses. If you are having problems with this then you
might want to send us a wireshark trace containing a session
that includes early media so that we could try and see where
the problem comes from.

Cheers, Emil

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

---------------------------------------------------------------------
To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net
For additional commands, e-mail: dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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#8

Hi Emil,

it works just fine now :slight_smile:

Cheers,
Fabio

···

Il giorno 22/ott/2010, alle ore 12.28, Emil Ivov ha scritto:

На 22.10.10 09:42, Fabio Telme написа:

Hi Emil

i've just installed the latest build but the double ringtone still happen
hope it can be fixed :slight_smile:

Well it appears that the server is sending the 180 after the 183 and not
the other way around.

Should be OK in build 3037.

Cheers,
Emil

Thanks
Fabio
Il giorno 22/ott/2010, alle ore 01.47, Emil Ivov ha scritto:

Hey Fabio,

На 21.10.10 23:58, Fabio Telme написа:

Hi Emil,

sorry for my earlier email, but i did not noticed that SC plays both
ring,

Should be alright now.

Cheers,
Emil

the local and remote at the same time, Marc how is it for You
?

Thanks Fabio

Il giorno 21/ott/2010, alle ore 23.23, Emil Ivov ha scritto:

Hey Fabio,

In case you hadn't noticed, this is about the resolution of the
"early media" problem that you reported earlier, and that we have
now fixed:

https://sip-communicator.dev.java.net/servlets/ReadMsg?list=dev&msgNo=9699

Cheers,

Emil

На 21.10.10 14:55, Emil Ivov написа:

На 21.10.10 14:44, Fabio Telme написа:

Hi Emil,

thanks very much for the answer, in attach the cap file i made
from my PC, i see in the capture that there are both 183 and
183 SDP signals, i'm not sure at this point if the problem
comes from the SIP Server, but other clients always reproduce
the correct session progress so this is the first time i
experience this problem. Hope you can find out the reason :slight_smile:

You server doesn't seem to be sending any RTP traffic before the
200 OK so there's simply no early media for SC to play.

You'd have to check the server configuration and logs to
understand why this is happening. You can also try and see
whether there are any differences in the dump when doing the same
thing with clients that receive the early media in your case.

Hope this helps, Emil

Many Thanks Fabio

Il giorno 21/ott/2010, alle ore 12.55, Emil Ivov ha scritto:

Hey Fabio,

На 18.10.10 22:37, Fabio Telme написа:

Hi

I'm testing several Sip Server and i noticed that for the
call progress is enable only the 180SDP method that
reproduce local ringback for alerting the call progress.
SIP calls to PSTN numbers should be able to accept the
183SDP signal to reproduce remote early tone to alert call
progress. So when call to PSTN, if the remote party respond
with 183SDP you should play the remote early media and not
the local ringback.

SIP Communicator already supports 183 Session Progress
responses. If you are having problems with this then you
might want to send us a wireshark trace containing a session
that includes early media so that we could try and see where
the problem comes from.

Cheers, Emil

---------------------------------------------------------------------

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For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

-- Emil Ivov, Ph.D. 67000
Strasbourg, Project Lead France
SIP Communicator emcho@sip-communicator.org
PHONE: +33.1.77.62.43.30 http://sip-communicator.org
FAX: +33.1.77.62.47.31

---------------------------------------------------------------------

To unsubscribe, e-mail: dev-unsubscribe@sip-communicator.dev.java.net

For additional commands, e-mail:
dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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For additional commands, e-mail: dev-help@sip-communicator.dev.java.net

--
Emil Ivov, Ph.D. 67000 Strasbourg,
Project Lead France
SIP Communicator
emcho@sip-communicator.org PHONE: +33.1.77.62.43.30
http://sip-communicator.org FAX: +33.1.77.62.47.31

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