Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Henning Westerholt]

Closed #1173.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Henning Westerholt]

Close this one - please submit your code as a new pull request if you are still 
interested on working on this.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

Hey, thanks for taking care of this. Will be happy to have some of my code
finally submitted. I also sawed it out and tested it every which way but
loose. Don.

On Fri, Nov 2, 2018 at 1:05 AM ng-voice GmbH 
wrote:

> I've reviewed the code and tested it intensively. I will make patch of it
> soon. Due to a busy schedule, this patch did not make it upstream yet
>
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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[ng-voice GmbH]

I've reviewed the code and tested it intensively. I will make patch of it soon. 
Due to a busy schedule, this patch did not make it upstream yet

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Henning Westerholt]

I did a quick search for some of the functions from the code block above, did 
not found them in git master.

@ngvoice - did you managed to commit this already? 

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

Hi guys, I am no longer at Verizon, but, can still help with this issue if
needed. Let me know. I do look forward to working on Kamailio again
sometime in the future. Don.

On Sat, Mar 3, 2018 at 12:39 PM, henningw  wrote:

> Hi Carsten, was this already commited? Somehow I did not found it, maybe I
> did not looked carefully enough.
>
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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[henningw]

Hi Carsten, was this already commited? Somehow I did not found it, maybe I did 
not looked carefully enough.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

Hey, thank you VERY much Carsten, that is really appreciated. Sorry that I
did not have time to take care of it. Don

On Tue, Sep 12, 2017 at 9:27 AM, ng-voice GmbH 
wrote:

> Hi,
>
> i've reviewed the code and made some minor improvements. I will commit a
> slightly modified version later this week.
>
> Thanks,
> Carsten
>
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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[ng-voice GmbH]

Hi,

i've reviewed the code and made some minor improvements. I will commit a 
slightly modified version later this week.

Thanks,
Carsten

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

Hey, very sorry, I got assigned another project and have not been able to
get back to this. Hopefully I will have some time soon. I think it will be
a good update that will avoid future problems. Don.

On Sun, Sep 3, 2017 at 3:51 AM, Daniel-Constantin Mierla <
notificati...@github.com> wrote:

> Any chance for a PR sometime soon, so this item can be closed here?
>
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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Daniel-Constantin Mierla]

Any chance for a PR sometime soon, so this item can be closed here?

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

@miconda, thanks, that is exactly what I have now, so, will submit it when I 
can. Below is the code that I wrote with a new function named "std_tel2sip()", 
as a preview I used most of the old procedure where possible, and tried to keep 
the same style.

I have been given a task with a short deadline, so, will not be able to work on 
this anymore right now. I will get back to it as soon as possible.

Don.

```
/*
 * Compare function to sort tel: uri options acording to standard
 * before inserting into sip: uri
 *
 * See "RFC 3261 SIP: Session Initiation Protocol June 2002"  
 * 19.1.6 Relating SIP URIs and tel URLs
 */
typedef struct
{
char *name;
char *value;
} tel_param_t;
#define MAX_TEL_PARAMS (10)

int compare_tel_options(const void *v1, const void *v2)
{
tel_param_t *p1 = (tel_param_t *) v1;
tel_param_t *p2 = (tel_param_t *) v2;

if (0 == strcasecmp(p1->name, "isdn-subaddress"))
{
return -1;
}
else if (0 == strcasecmp(p2->name, "isdn-subaddress"))
{
return 1;
}
else if (0 == strcasecmp(p1->name, "post-dial"))
{
return -1;
}
else if (0 == strcasecmp(p2->name, "post-dial"))
{
return 1;
}
else
{
return strcasecmp(p1->name, p2->name);
}
}

/*
 *   Remove visual separators from the phone number 
 *   Assume it has been validated as a number containing 
 *   ONLY leading '+', digits, and visual separators.
 */
static void remove_visual_separators_from_phone(char *p)
{
char *p2;
p2 = p;
while (*p != '\0')
{
 /* Skip all visual separators */
 while ((*p != '\0') && ((*p == '.') || (*p == '-') || (*p == 
'(') || (*p == ')')))
 {
 p++; 
 }
 *p2 = *p; /* Until the first visual separator, these both 
point to  the same place. */
   /* but, more efficient to just do it than an if 
statement each time. */
 /* If we arrived at a terminator in the inner loop, time to 
exit */
 if (*p == '\0') return;
 /* Now we increment both pointers. */
 p++;
 p2++;
}
*p2 = '\0';  /* Make sure that the string is terminated after the last 
valid digit. */
}

/*
 * Check if this is a phone number.
 *   Assume possible leading '+'
 *   Assume separators '.', '-', '(', or ')' could be present.
 */
static int is_number(const char *p)
{
if (*p == '+') p++;
while (*p != '\0')
{
 if ((!isdigit(*p)) && (*p != '.') && (*p != '-') && (*p != 
'(') && (*p != ')')) return 0;
 p++;
}
return 1;
}

/*
 * Converts URI, if it is tel URI, to SIP URI.  Returns 1, if
 * conversion succeeded or if no conversion was needed, i.e., URI was not
 * tel URI.  Returns -1, if conversion failed.  Takes SIP URI hostpart from
 * second parameter and (if needed) writes the result to third parameter.
 * This one attempts to be standards compliant and sort tel: uri parameters
 * copied to the sip: uri in the manner defined in the standard. It also 
 * deletes the "phone-context" parameter if it is a domain, and, takes visual
 * separators from the "phone-context" parameter if it is a telephone number.
 */
int std_tel2sip(struct sip_msg* _msg, char* _uri, char* _hostpart, char* _res)
{
str uri, hostpart, tel_uri, sip_uri;
char *at;
int i, j, in_tel_parameters = 0;
pv_spec_t *res;
pv_value_t res_val;

/* get parameters */
if (get_str_fparam(&uri, _msg, (fparam_t*)_uri) < 0) {
LM_ERR("failed to get uri value\n");
}
if (get_str_fparam(&hostpart, _msg, (fparam_t*)_hostpart) < 0) {
LM_ERR("failed to get hostpart value\n");
}
res = (pv_spec_t *)_res;

/* check if anything needs to be done */
if (uri.len < 4) return 1;
if (strncasecmp(uri.s, "tel:", 4) != 0) return 1;

/* reserve memory for clean tel uri */
tel_uri.s = pkg_malloc(uri.len+1);
if (tel_uri.s == 0) {
LM_ERR("no more pkg memory\n");
return -1;
}

/* Remove visual separators before converting to SIP URI. Don't remove
 * visual separators in TEL URI parameters (after the first ";") */
for (i=0, j=0; i < uri.len; i++) {
if (in_tel_parameters == 0) {
if (uri.s[i] == ';')
in_tel_parameters = 1;
}
if (in_tel_parameters == 0) {
if ((uri.s[i] != '-') && (uri.s[i] != '.') &&
(uri.s[i] != '(') && (uri.s[i] != ')'))
tel_uri.s[j++] = tolower(uri.

Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Daniel-Constantin Mierla]

I would also go straight for a function that give the final result. Having a 
two step approach is not useful and adds to complexity of the config. The 
existing function should be used by those that fit their need when they know 
what they expect. The new one, by those that want to ensure the result is 
standards compliant, even for unknown sources.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

@juha-h, @miconda ,

On making a "normalizing" function.  I could do that to remove the visual 
separators from the "phone-context" field when it is a phone number, and, also, 
pre-sort the parameters in the tel: uri, so that when the current (light 
weight) function pulls them, the parameters are already in the sorted order, 
and there are no visual separators in the "phone-context" field. 

A little more tricky, I could also delete the "phone-context" field from the 
tel: uri when it is a domain, assuming that the host name was already pulled 
out. But, if this was done before domain was extracted, it would cause an error 
(basically losing the domain). Currently, the domain in the "hostpart" variable 
is inserted after the "@" sign, and the contents in the "phone-context" field 
is inserted as is. 

Also, in this case, there can be no logic in deciding which domain to select 
(the one in the "hostpart", or the one in the "phone-context"  It seems that 
the hostpart is currently pulled using cscf_get_realm_from_uri(), so, 
theoretically, the variable "hostpart" and the contents of "phone-context" 
should always be the same.

Ok, in any case, I think it would be a little confusing to have a pre-filter 
that creates a special format of a tel: uri such that an existing translator 
will end up creating a standards compliant sip: uri. 

I think it would be best to give a simple/fast version and a standards 
compliant version.

The existing function does try to do part of the standards in that it converts 
everything to lower case, and, removes visual separators from the phone number 
digits before the first semi-colon.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Juha Heinanen]

Donald Carr writes:

> Are you interested in the fix? I agree it is a bit esoteric, but, I
> think worth it to help reduce problems in the future. Especially
> removing the double domain name in the sip uri, and removing visual
> separators from the phone-context when it is a global number. Well,
> the parameters really should be sorted to meet the spec, but, not sure
> if any program would ever have a problem if they were not sorted. It
> is nice if equivalent uris look exactly the same.

How about making a new function that "normalizes" tel uri (does sorting,
etc.)?  That function would then be called before calling simple tel2sip
if all the fancy stuff is needed.

-- Juha

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Daniel-Constantin Mierla]

@doncarr - yes, make a pull request, I suggested to be a new function so the 
lighweight version stays around and can be used by those that know it fits 
their needs.

FYI: `kamailio-sync` is an account associated with sr-dev mailing list just in 
case someone replies there, instead of using the github webforms for issues. In 
the above case, it was @juha-h responding on mailing list (as shown in the 
signature when expanding the dots at the end of comments). 

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

@kamailio-sync, @miconda,

Are you interested in the fix? I agree it is a bit esoteric, but, I think worth 
it to help reduce problems in the future. Especially removing the double domain 
name in the sip uri, and removing visual separators from the phone-context when 
it is a global number. Well, the parameters really should be sorted to meet the 
spec, but, not sure if any program would ever have a problem if they were not 
sorted. It is nice if equivalent uris look exactly the same.

Please let me know, Don.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

I have a fix just about ready for your review.  As noted, there can be some 
weird things in the telephone world!

There are a couple of not well defined issues. They more or less state that the 
objective of the sip: uri, is that two equivalent uris will always be the 
equivalent string, thus the need to sort the parameters and converting to lower 
case. That much is very clear. But, applying this same reasoning to the 
"phone-context" parameter, if it is a domain, then, it should be moved after 
the '@' symbol and not repeated as a "phone-context" parameter, as to have an 
equivalent string to one without the phone-context parameter.  

For example, the following two are obviously equivalent, but, are not the same 
string:

sip:491234567891;phone-context=ims.tel@ims.tel.org;user=phone
sip:491234567...@ims.tel.org;user=phone

Also, they do not specifically state that visual separators should be removed 
from the "phone-context" parameter when it is a global phone number prefix, 
but, without removing them, we would again have the problem of equivalent sip 
uris not being equivalent strings. 

For example, the following two are obviously equivalent, but, not the same 
string:

sip:1234567;phone-context=+52...@ims.tel.org;user=phone
sip:1234567;phone-context=+5...@ims.tel.org;user=phone

If that all sounds good to you, I will submit a fix after more testing.

I have not messed with checking for clearly bad use of visual separators, I 
just silently remove all visual separators wherever they appear, as was done 
with the existing code.

Please let me know if you want any changes to the reasoning. You could also 
give me example tel: uris you would like me to test before submitting the fix. 

Thanks for everything! 

Don.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Juha Heinanen]

Donald Carr writes:

> At the link below, you can go to 19.1.6. It is a very strange ordering
> giving priority to two parameters that should be first, if present.
> 
> https://www.ietf.org/rfc/rfc3261.txt
> 
> 19.1.6 Relating SIP URIs and tel URLs
> . . . . 
> ```
>To mitigate this problem, elements constructing telephone-subscriber
>fields to place in the userinfo part of a SIP or SIPS URI SHOULD fold
>any case-insensitive portion of telephone-subscriber to lower case,
>and order the telephone-subscriber parameters lexically by parameter
>name, excepting isdn-subaddress and post-dial, which occur first and
>in that order.  (All components of a tel URL except for future-
>extension parameters are defined to be compared case-insensitive.)
> ```

OK, telephone world is very weird.

-- Juha


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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

Actually, I intended to use most of the existing function, just inserting a 
sort, and, removal of duplicate domain. But, yes, it could be a new function, 
leaving the existing one exactly as-is. 

It would also be nice for the update to be active without having to edit the 
configuration file. I will leave it totally up to you on the best way to 
integrate.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Daniel-Constantin Mierla]

@doncarr - ti can be also a new function, like tel_to_sip_uri(), so if one 
wants to use the old/lightweight version can still do it. I guess the new 
function is not going to share too much code with the old onw.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

At the link below, you can go to 19.1.6. It is a very strange ordering giving 
priority to two parameters that should be first, if present.

https://www.ietf.org/rfc/rfc3261.txt

19.1.6 Relating SIP URIs and tel URLs
. . . . 
```
   To mitigate this problem, elements constructing telephone-subscriber
   fields to place in the userinfo part of a SIP or SIPS URI SHOULD fold
   any case-insensitive portion of telephone-subscriber to lower case,
   and order the telephone-subscriber parameters lexically by parameter
   name, excepting isdn-subaddress and post-dial, which occur first and
   in that order.  (All components of a tel URL except for future-
   extension parameters are defined to be compared case-insensitive.)
```


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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Juha Heinanen]

Donald Carr writes:

> @miconda, I have written a routine that puts the parameters for the
> tel: uri into an array with key/value separated, then uses qsort to
> sort. With this,  they can be put into the sip: uri in correct sorted
> order. The sort is very esoteric, specific to the requirements for
> sip: uri when converting from a tel: uri ...

Could you give an example where sorting is needed and a pointer to
a relevant RFC?

-- Juha

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

@miconda, I have written a routine that puts the parameters for the tel: uri 
into an array with key/value separated, then uses qsort to sort. With this,  
they can be put into the sip: uri in correct sorted order. The sort is very 
esoteric, specific to the requirements for sip: uri when converting from a tel: 
uri, so, I would suggest we add it to the existing conversion routine. Though, 
it would be very easy to make it a separate callable script called before the 
existing routine, to sort the tel: uri. Also, if inside the current routine, at 
the same time that we have the parameters split up, we can easily check if 
phone-context is present, and, if it is a domain, we can delete the 
phone-context parameter, and, move the domian to the standard place after the @ 
symbol.

Please let me know how we can move forward. I can create a modified tel2sip() 
routine with the sorting of tel: parameters and a check to eliminate duplicate 
domains. Then, do testing here, and, finally send to you for your perusal 
before accepting.

Thank you very much!

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

@miconda, thanks very much! I will take a stab at an improvement and submit, 
starting with sorting the parameters according to the spec. This is good for me 
to understand the spec and kamailio code, and a small part of, over time, 
making kamailio better so that it can/will be used in more places.

I do understand that there are trade-offs in enforcing esoteric details of the 
spec, and, in any case, it would be impossible (and not even desirable) to try 
to enforce everything.

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Re: [sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Daniel-Constantin Mierla]

I expect that tel2uri() was implemented based on the needs of initial developer 
as documented at:

  * 
https://www.kamailio.org/docs/modules/devel/modules/siputils.html#siputils.f.tel2sip

It can be improved if one needs/wants more.

Note also that specs are not implemented ad-litteram in a single function. The 
design of the application has been from the beginning to offer small tools 
(functions/statements) that can be blended in config script to get what you 
need. There are many functions and transformations that can help you 
replace/remove unwanted characters from user part of R-URI, such as 
subst_user(...), {re.subst} or {s...} transformations, etc. With all these I am 
pretty sure you can transfrorm tel to sip uri as per specs without using 
tel2sip() function.

To be clear: I am not against enhancing tel2sip() to do more that it does now, 
but wanted to highlight that many spec requirements can be done with a 
combination of functions and expressions. I am not the author of tel2sip() and 
actually I haven't used it many times. I do URI normalization with script 
operations.

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[sr-dev] [kamailio/kamailio] Incorrect translation of tel: uri to sip: uri in tel2sip() (#1173)

[Donald Carr]

### Description



I received a wireshark trace from a team member showing that a tel: uri was 
translated to a sip: uri in a very strange (invalid) way:

INVITE tel:491234567891;phone-context=ims.mnc001.mcc001.3gppnetwork.org
 
Was translated to:

INVITE  
sip:491234567891;phone-context=ims.mnc001.mcc001.3gppnetwork@ims.mnc001.mcc001.3gppnetwork.org;user=phone



### Troubleshooting

I first thought this was definitely an error, and only the phone number should 
be copied to the sip: uri. But, on further reading of the specification, 
(https://www.ietf.org/rfc/rfc3261.txt), I see that it is in fact legal to leave 
all parameters in. Though, when the phone-context parameter is simply a domain, 
it would at least seem the cleanest thing would be to remove it when the exact 
same domain is also inserted after the @ in the sip message. 

So, looking to find where the message is translated, I saw in the scscf.cfg 
configuration file that there is a translation specified. 

I found that "tel2sip" is mapped to the C function of the same name tel2sip()

In this function, the translation will do exactly what I found in the trace, in 
other words, it will leave in all parameters separated by semi-colon, inserted 
before the @ symbol.

But, reading the spec, I see that the spec is not being followed in any case. 
The spec states that all parameters included when translating, must be switched 
to alphabetic order, with the exception that 
isdn-subaddress and post-dial, must occur first and in that order.  There is no 
code in tel2sip() to perform that sorting.

Also, the parser removes '(', ')', '.', and '-' in the phone number, as 
specified, but, these same characters could be scattered anywhere in invalid 
places, and they are just silently removed, '+' can scattered anywhere and it 
is just left alone, Other characters can be scattered anywhere and they will 
just be left as-is, no error. So, I feel there should be some minimal checking 
of the phone number, trying to follow the spec (if we could just execute the 
BNF from the spec!).

 Reproduction


This can be reproduced by sending a tel: uri with parameters inserted separated 
by semicolon.

 Debugging Data



```
(none)
```

 Log Messages



```
(none)
```

 SIP Traffic



```
See attached file
```

### Possible Solutions



The solution would be to modify tel2sip(), adding the sorting of parameters, 
putting the two exceptions first if they exist, before inserting back into the 
string.

We should also throw errors on obvious bad formatting and ilegal characters, 
etc, in the input tel: uri

Should we remove the "phone-context" if it is a domain and also exactly the 
same as the domain inserted after the @? 

### Additional Information

  * **Kamailio Version** - output of `kamailio -v`
```
version: kamailio 5.0.0-dev5 (x86_64/linux) da4ef4-dirty
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS, 
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC, 
F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, 
USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, 
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: da4ef4 -dirty
compiled on 10:37:40 May 12 2017 with gcc 5.4.0
```
SIP spec:
https://www.ietf.org/rfc/rfc3261.txtWireshark output:

![wireshark-tel2sip](https://user-images.githubusercontent.com/4886876/27705203-cfd3c150-5cd2-11e7-9f77-1664a51e7ac3.png)
```
In RFC 3966, Tel URI format:
 
   The "tel" URI has the following syntax:
 
   telephone-uri= "tel:" telephone-subscriber
   telephone-subscriber = global-number / local-number
   global-number= global-number-digits *par
   local-number = local-number-digits *par context *par
   par  = parameter / extension / isdn-subaddress
   isdn-subaddress  = ";isub=" 1*uric
   extension= ";ext=" 1*phonedigit
   context  = ";phone-context=" descriptor
   descriptor   = domainname / global-number-digits
   global-number-digits = "+" *phonedigit DIGIT *phonedigit
   local-number-digits  =
  *phonedigit-hex (HEXDIG / "*" / "#")*phonedigit-hex
   domainname   = *( domainlabel "." ) toplabel [ "." ]
   domainlabel  = alphanum
  / alphanum *( alphanum / "-" ) alphanum
   toplabel = ALPHA / ALPHA *( alphanum / "-" ) alphanum
   parameter= ";" pname ["=" pvalue ]
   pname= 1*( alphanum / "-" )
   pvalue   = 1*paramchar
   paramchar= param-unreserved / unreserved / pct-encoded
   unreserved   = alphanum / mark
   mark = "-" / "_" / "." / "!" / "~" / "*" /
  "'" / "(" / ")"
   pct-encoded  = "%" HEXDIG HEXDIG
   param-unreserved