2009/7/16 Vikram Chhibber vikram.chhib...@gmail.com:
It does not make sense to transaction from active to pending. RFC 3857
also does not show this transaction.
Why not move from active to terminated with reason rejected? Alice
UA can show Bob as offline always. Which IMO, is modest approach
2009/7/16 Vikram Chhibber vikram.chhib...@gmail.com:
It does not make sense to transaction from active to pending. RFC 3857
also does not show this transaction.
Why not move from active to terminated with reason rejected? Alice
UA can show Bob as offline always. Which IMO, is modest approach
I think most of the times it depends upon the implementation.
In one of the product which I had worked on, this was one of the configurable
options to be turned on if we have to detect some kind of rogue packets.
There will be different checks to be made to consider the packet as the rogue
and
2009/7/16 Iñaki Baz Castillo i...@aliax.net:
2009/7/16 Vikram Chhibber vikram.chhib...@gmail.com:
It does not make sense to transaction from active to pending. RFC 3857
also does not show this transaction.
Why not move from active to terminated with reason rejected? Alice
UA can show Bob as
Hi all,
is it a problem, when a RTP-stream during a call is replaced by another
RTP-stream?
Let's assume, stream with SSRC1 is arriving at a UAC
(100.11.12.13:12345). Now, the stream stops (without any SIP-signalling
like Re-INVITEs, etc.) and a stream with SSRC2 is now arriving at
Hi All,
Is there any SIPSoft pohne which support G729AB codec?
I tried googling but I couldn't succeed on same.
Please do let me know if any info on same
Thanks and Rgds
Sudhir
Yahoo! recommends that you upgrade to the new and safer Internet Explorer
8.
2009/7/16 Sudhir Kumar Reddy sudhir_kuma...@yahoo.co.in:
Hi All,
Is there any SIPSoft pohne which support G729AB codec?
This is not a question for this list.
However, you won't find a free softphone with G729 codec (it requires
royalties payment).
--
Iñaki Baz Castillo
i...@aliax.net
Sudhir Kumar Reddy wrote:
Is there any SIPSoft pohne which support G729AB codec?
I tried googling but I couldn't succeed on same.
Please do let me know if any info on same
The purpose of this mailing list is to discuss SIP protocol mechanics,
formal behaviour per IETF standards, and
Hi,
but is there a standard defining the behaviour? I guess, there is no
need to compare SSRC of every incoming RTP-packet on the client (Well -
security reasons as RTP-spoofing left beside). I am looking for a
quotable reference regarding the SSRCs inside of RTP-streams managed by
SIP/SDP
Hi,
Outbound proxy is a 100% SIP Proxy.
The distinction is based on the logical functions it performs.
It is the intermediate SIP node which needs to be included
on the signalling path in order to get access to SIP services.
It is usually pointed be the predefined Route set.
Kind regards,
-
Hi,
but is there a standard defining the behaviour? I guess,
there is no need to compare SSRC of every incoming RTP-packet
on the client (Well - security reasons as RTP-spoofing left
beside). I am looking for a quotable reference regarding the
SSRCs inside of RTP-streams managed by
A participant need not use the same SSRC identifier for all the RTP
sessions in a multimedia session; -- RFC 1918, Section 3.
In practice, many devices do not expect the SSRC to change routinely.
Heuristically, in an ordinary phone call, a changing SSRC may indicate
that two streams are
Hi all,
is it a problem, when a RTP-stream during a call is replaced by another
RTP-stream?
Let's assume, stream with SSRC1 is arriving at a UAC
(100.11.12.13:12345). Now, the stream stops (without any SIP-signalling
like Re-INVITEs, etc.) and a stream with SSRC2 is now arriving at
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