20 Feb
2005
20 Feb
'05
4:35 a.m.
Hello!
I'm implementing some VoIP system with SER+Asterisk. I've already
implemented many features, but all of them was tested only in non-NAT
environments. Now, at final-alpha stage (before starting massive beta
tests) I found big (as for me) problem with NAT... Now I need a working
example of ser.cfg (it will be really good) or, simply, to talk with
someone, who realized such scheme...
My current problem is with ua-to-ua calls between my users...
I have a following call routing scheme...
UA1->...->SER->Asterisk->SER->...->UA2...
Description: UA1 calls to some number, ser routes this call to asterisk,
asterisk resolves that number to some account (login name, that being
used by users to login to ser) and makes call to sip/ser/user_name...
and ser routes this call for user with such login...
I have strange problems with natted UAs...
Please, send me a working example of similiar schemes
(ser+asterisk+nat)... or help me by a word ;-)
Great thanks!
--
/Scoundrel
22 Feb
22 Feb
8:19 a.m.
New subject: [Serusers] SER+Asterisk+NAT. Need a working example or need to talk about it.
I just fininsh mediaproxy+ser, and I also want to do the same plan as you.
Can you give me some example config (ser.cfg & asterisk's config files)?
On Sun, 20 Feb 2005 04:35:02 +0200, Alexey N. Kovyrin @ Home
<alex(a)home.kovyrin.net> wrote:
Hello!
I'm implementing some VoIP system with SER+Asterisk. I've already
implemented many features, but all of them was tested only in non-NAT
environments. Now, at final-alpha stage (before starting massive beta
tests) I found big (as for me) problem with NAT... Now I need a working
example of ser.cfg (it will be really good) or, simply, to talk with
someone, who realized such scheme...
My current problem is with ua-to-ua calls between my users...
I have a following call routing scheme...
UA1->...->SER->Asterisk->SER->...->UA2...
Description: UA1 calls to some number, ser routes this call to asterisk,
asterisk resolves that number to some account (login name, that being
used by users to login to ser) and makes call to sip/ser/user_name...
and ser routes this call for user with such login...
I have strange problems with natted UAs...
Please, send me a working example of similiar schemes
(ser+asterisk+nat)... or help me by a word ;-)
Great thanks!
--
/Scoundrel
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10:06 a.m.
New subject: [Serusers] SER+Asterisk+NAT. Need a working example or need to talk about it.
Charles Wang wrote:
I just fininsh mediaproxy+ser, and I also want to do
the same plan as you.
Can you give me some example config (ser.cfg & asterisk's config files)?
Now, I'm cleaning up my configs for more readability... I'll send it to
you for 1-2 days...
But, I can say, this scheme is working now. And it's good.
That's why I really want to participate in "Best practices" initiative -
I've seen, that there si almost no docs about ser...
--
/Scoundrel
11:12 a.m.
New subject: [Serusers] FYI: Reference document
To All,
Paul, Simon, and I have started the work on an initial draft of a ser
reference design document. We are trying to nail down the "must-have's" and
get a first draft ready. We will leave room for contributions on non-core
(i.e. not necessary for 80% of users) configs/elements.
I suggest that Alexey Kovyrin and others who would like to contribute
with a section to post an email with a statement describing the topic, as
well as a short description of what he/she feels should be covered. We will
coordinate and make sure that there is no overlap.
Thanks,
Greger
3:47 p.m.
New subject: [Serusers] FYI: Reference document
Excellent news: Just adding my topic, in addition can I suggest that the
subject line for topic posts to be changed to SERDOC (that way i wont
lose it in all my posts :-))
I dont know the correct format that you want this is, if it needs
changing let me know
Topic: RTP stream handling and PSTN billing
Since alot of users will be using pstn termination, I guess a small
section covering best practice for pstn termination and billing.
a) best way of doing billing, always been apopular question, always
answered with USE b2bua
b) The above solution is fine, but its does not describe the probs that
may occur with scaling issues if you are handling the media stream
c) Since asterisk will be included in the voicemail section, I guess a
addon here about handling outbound pst, together with the CDR rating
proggies that it has might be useful (I am not sure if this is skewing
too much towards asterisk)
d) call scenraios, sip --> sip (both behind nat)
sip ---> sip ( caller or callee behind nat)
sip ---> pstn ( caller behind/not behind nat)
pstn ----> sip (callee behind/not behind nat)
Each of these scenarios would affect the setup, in that some are just
being handled by the mediaproxy/rtpproxy others wont need that, but the
rtp stream would need to pass through a b2bua to aid billing....if your
doing prepaid
e) sip ---> sip of other provider...SIP interconnect, I guess should also
be included, possibly with a standard prefix for each provider (there
was a post on this list about standard prefixes, or a maintainer)
If more/less info is needed let me know, currently I am working on the
setup via asterisk for outbound pstn simply for prepaid, hence once I
have it up, I could give the docs for the rating stuff...
Iqbal
On 2/22/2005, "Greger V. Teigre" <greger(a)teigre.com> wrote:
To All,
Paul, Simon, and I have started the work on an initial draft of a ser
reference design document. We are trying to nail down the "must-have's" and
get a first draft ready. We will leave room for contributions on non-core
(i.e. not necessary for 80% of users) configs/elements.
I suggest that Alexey Kovyrin and others who would like to contribute
with a section to post an email with a statement describing the topic, as
well as a short description of what he/she feels should be covered. We will
coordinate and make sure that there is no overlap.
Thanks,
Greger
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4 comments
5 participants
participants (5)
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Alexey N. Kovyrin ＠ Home -
Alexey N. Kovyrin ＠ Home -
Charles Wang -
Greger V. Teigre -
Iqbal