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
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@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
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
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...
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
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@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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