Module: kamailio
Branch: master
Commit: 1cfd694e3bad62e9c4fc31073fcebe7707c5968c
URL: https://github.com/kamailio/kamailio/commit/1cfd694e3bad62e9c4fc31073fcebe7…
Author: Daniel-Constantin Mierla <miconda(a)gmail.com>
Committer: Daniel-Constantin Mierla <miconda(a)gmail.com>
Date: 2023-10-30T13:27:49+01:00
secsipid: docs for secsipid_sign_prvkey()
---
Modified: src/modules/secsipid/doc/secsipid_admin.xml
---
Diff: https://github.com/kamailio/kamailio/commit/…
[View More]1cfd694e3bad62e9c4fc31073fcebe7…
Patch: https://github.com/kamailio/kamailio/commit/1cfd694e3bad62e9c4fc31073fcebe7…
---
diff --git a/src/modules/secsipid/doc/secsipid_admin.xml b/src/modules/secsipid/doc/secsipid_admin.xml
index 97306c649b3..7ea74f525a0 100644
--- a/src/modules/secsipid/doc/secsipid_admin.xml
+++ b/src/modules/secsipid/doc/secsipid_admin.xml
@@ -460,6 +460,39 @@ request_route {
...
}
...
+</programlisting>
+ </example>
+ </section>
+ <section id="secsipid.f.secsipid_sign_prvkey">
+ <title>
+ <function moreinfo="none">secsipid_sign_prvkey(sheaders, spaypload, keyData)</function>
+ </title>
+ <para>
+ Build Identity value using the private key given by "keyData" to sign the JWT body.
+ The sheaders and spayload have to be string representation of JSON
+ headers and payload to be signed. On success, the Indentity value is
+ stored in variable $secsipid(val). It also sets $secsipid(ret) to
+ the return value of the libsecsipid functions.
+ </para>
+ <para>
+ The parameters can contain pseudo-variables.
+ </para>
+ <para>
+ This function can be used from ANY_ROUTE.
+ </para>
+ <example>
+ <title><function>secsipid_sign_prvkey</function> usage</title>
+ <programlisting format="linespecific">
+...
+request_route {
+ ...
+ if(secsipid_sign_prvkey("_JSON_HEADERS_", "_JSON_PAYLOAD_",
+ "_PRIVATE_KEY_")) {
+ xinfo("Identity value: $secsipid(val)\n");
+ }
+ ...
+}
+...
</programlisting>
</example>
</section>
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hello
im coming back to ask on this forum, sorry if im at the wrong place, but i guess there are some sip experts around here (if it's not the right place, tell me where...)
i was wondering, what about the public enum usage for sip networks?
i mean, is there any type of public networks, known for having an opened/accessible users scheme?
eg : here in france there are linphone and ippi, and others (providing software+service)
both, offers both software+service, but can be dissociated
…
[View More]means its possible (normally) to use another sip software with a provider, etc
the idea is, there is what i think called a DID/URI, whom looks like email principle :
user(a)domain.com ; but in a way, it's a bit similar for xmpp and sip : alice(a)first.com could email, call or message bob(a)second.net ; and normally, it's minded to be interoperable (or federated) on an opened-network circuit.
what i try to understand is :
URI or DID might be i guess, the user(a)domain.net.
however, if sip is really used today, it's on closed networks. I mean, using data, only whatsapp and others, I guess uses xmpp or sip, but not in an opened-way, means not +123475(a)whatsapp.org, what would be great to be communicate with, without having to install their app.
on ""real"" opened-community, such as callcentrix or voip.ms, it's the case : a user of a domain could reach another domain, using it's DID/URI email.
well, I found then that there is a kind of ENUM thing I discovered few days ago. instead of defunct iNum, where it was supposed to be as an online voip accessible and free from internet (like skype) new kind of voip network, ENUM looks like to be enclosed one. As operators mainly looks for ""security"" (or security of closed-business), i found those things :
https://wiki.freepbx.org/display/DIMG/SIP+Enum+Supporthttps://en.wikibooks.org/wiki/Voice_over_IP/ENUM_and_E146_Technologyhttps://www.lightreading.com/cable-technology/the-impact-of-enum-on-voiphttps://lafibre.info/images/peering/201506_efort_carrier_enum.pdf
so if i understand well, mainly operators took SIP protocol to create their own voIP network, to their susbcribers ; not bad if it wasnt totally-closed?
imho, i then discovered that all operators around the world have (for a part of them) a voip/sip account for all volte or even for business lines. My mind was, sip looks like to be great working. But what surprised me in the bad way : why isnt it possible, because it's 100% data usage, to call them directly from a common "voip/callcentric/linphone/other" sip account?
i guess it's a question of money, but i dont understand then why people dont go directly on a sip account if their computer is internet connected h24 or even LTE/5G running permanently on their phone : it would be double unlimited plus free of charge international calls. Know people will tell me "just it's whatsapp", but no : whatsapp brings just simplicity + zero rating in some countries, not privacy-compliant and opened and federated usage. am I wrong?
on my own, I would have a little question for the nerds or passionate, like me, to SIP protocol for opened-networks, to ensure it's possible to simple voip communicate with relatives, on the same scheme as email : just call a contact by it's user(a)domain.net, and it might work. How could you convince your relatives to go on a such scheme?
I mean, in EU we have several voip initiative, whom are fully accessible from worldwide callers, where in N/america there are voip.ms and callcentric, maybe others.. are those users only using it for pstn-paid calls? why not skype then?
in addition : is there a directory of opened-sip URI/DID/ENUM to be in touch with, to see what operators/voip service providers permits their users to be called from anywhere around the world?
the main advantage of this is like email (or xmpp) : just with a voip/sip software, + a sip account, calls to any other sip user around the world, is generally totally free and unlimited. Why dont people adopt it, instead of whatsapp&etc?
are they some opened-voip providers directory, where some individuals or organisations could be called from over the world, through sip?
do you know some organisation, NGO or companies whom can be called from just a opened-sip network provider?
thank you for reading, and much more if you share your thoughts
(and sorry for bad eng..)
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hello,
i have tried with a friend recently, im unsure if it's the proper place to talk about it, but it's one of the rare forums with a category focused about SIP/VoIP.
here it is :
gigaset c530ip, i guess it's among the most solt model from the last decade, from that manufacturer.
on frenchy telcos forums, several nerds also have it, so i guess that device is pretty well know. What is highly less known, is the gigaset.net voip network offered by the manufacturer. I was thinking it was …
[View More]running about xmpp protocol. Looks like it's sip only. Well, it started like :
1. i tried to see if it was usable, alive, for a several yrs old device. So i looked into the phone's system, and saw a kind of directory, really difficult to use nowadays, regarding our smartphone's new features : it's both slow and limited in entry's public directory. well, half-public : browse it requires to get a gigaset voip professional phone.
2. when i discovered that "worldwide" directory, i tried to "call" like that, registred accounts. Uh: all were busy/unactive, no one seems available. From family-like accounts, to SMB or others usages, hundred of "sip account" are still registred, and listed.
3. well, after few months of little investigation, with a friend whom gets a such similar phone, we tried to see if it was a out of order network, or just a dormant one with thousand of forgotten accounts. we were then surprised to see that the network was working well, able to get a (unsecured) sip/voip communication.
4. thus, device looks like to get released in retail too early to support s/zrtp. however, even if between both same models, from two different cities, with two different internet access, the communication was working, but from another type of phone/service, it's not at all : i tried from both linphone/ippi software/services. Each time, the gigaset rings, shows the incoming call alert, but even when pushing the green button to hang up, isnt able to start a new communication : it's blocked on the gigast device, while the other (not gigaset, linphone) phone says its entered in communication, with no sound at all, for a while (about a minute) before the communication gets stopped.
qusetion is : how could it be fixed? anyone get success to call a sip gigaset device on gigaset.net network, with a working communication?
here are various logs we went about to retrieve from a friends (from linphone app or device's base with tcpdump) are here :
https://e.pcloud.link/publink/show?code=kZaqksZwM00MtijVWhqlGajWiyL7RWmDwg7
i thank you vm for your help !
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"200,405,486" ??
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which responses are ??
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