[SR-Users] Re: STIR/SHAKEN

2024-01-24 Thread Anthony Wittig via sr-users
Hey everyone,

I've set it up so that when the invite comes into Kamailio I suspend the
transaction (save the transaction info in sht) and use dispatcher to send a
request to our NSS server. On the reply from NSS I throw the identity
header into a sht and continue the original transaction. Does that seem
like the right way to do it?

The next things I need to look at are:
* if first NSS server doesn't respond quickly, try the second (change some
timeouts and use dispatcher to try the backup)
* if both NSS servers aren't working, I still want to resume the original
call and let it through. Maybe I can keep track of the failures in a
sht and if I have two failures, resume the original invite without the
identity header.

Sound like an ok plan?
--
Anthony Wittig
Mango Voice Developer

>
> -- Forwarded message --
> From: Anthony Wittig 
> To: sr-users@lists.kamailio.org, david.villasmil.w...@gmail.com
> Cc:
> Bcc:
> Date: Tue, 23 Jan 2024 20:39:09 -0700
> Subject: [SR-Users] STIR/SHAKEN
> Thanks!
>
> This is what I've got so far:
>
> 1. Kamailio invite -> NSS
> 2. NSS 302 -> Kamailio
> 3. Kamailio ack -> NSS
>
> I can see the identity header on the 302 response in my onreply_route. How
> can I replay the same invite from step #1 with the identity header I get
> back in step #2? Should I be doing something with a transaction on step #1,
> maybe start a branch that goes out to NSS and when it comes back resume the
> transaction (these are all words I've heard smart people say, but I don't
> fully understand them...).
>
> Any pointers would be greatly appreciated, thanks!
> --
> Anthony Wittig
> Mango Voice Developer
>
> -- Forwarded message --
> From: David Villasmil 
> To: "Kamailio (SER) - Users Mailing List" 
> Cc:
> Bcc:
> Date: Tue, 23 Jan 2024 14:56:52 +0100
> Subject: [SR-Users] Re: STIR/SHAKEN
> Ditto, i do first approach.
>
> Regards,
>
> David Villasmil
> email: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Tue, Jan 23, 2024 at 2:38 PM Alex Balashov via sr-users <
> sr-users@lists.kamailio.org> wrote:
>
>> These are all valid approaches, depending on preference. Catching 302s
>> and extracting exactly the desired info might be easier with Kamailio, so I
>> suppose, ceteris paribus, I'd recommend the first one.
>>
>> Dispatcher is a great approach!
>>
>> > On 23 Jan 2024, at 00:01, Anthony Wittig via sr-users <
>> sr-users@lists.kamailio.org> wrote:
>> >
>> > Hello,
>> >
>> > We're attempting to add identities to our invites. I believe we want to
>> send our invites to the NSS, and it'll reply with a 302'd invite with an
>> identity header. We currently use FreeSWITCH for dialplans and media and
>> Kamailio as our SBC.
>> >
>> > Should we be trying for something like: FreeSWITCH -> Kamailio -> NSS
>> (302) -> Kamailio -> PSTN
>> >
>> > or would it make more sense to do: FreeSWITCH -> NSS (302) ->
>> FreeSWITCH -> Kamailio -> PSTN
>> >
>> > Assuming we wanted Kamailio -> NSS, is using dispatcher the right
>> approach? We currently use it for our two Kamailios and our FreeSWITCHs.
>> We're going to have two NSS servers, one for failover.
>> >
>> > Any thoughts would be greatly appreciated, thanks!
>> >
>> > --
>> > Anthony Wittig
>> > Mango Voice Developer
>> > __
>> > Kamailio - Users Mailing List - Non Commercial Discussions
>> > To unsubscribe send an email to sr-users-le...@lists.kamailio.org
>> > Important: keep the mailing list in the recipients, do not reply only
>> to the sender!
>> > Edit mailing list options or unsubscribe:
>>
>> --
>> Alex Balashov
>> Principal Consultant
>> Evariste Systems LLC
>> Web: https://evaristesys.com
>> Tel: +1-706-510-6800
>>
>
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[SR-Users] Re: 412 Conditional request failed

2024-01-24 Thread satyaprakash ch via sr-users
Hi,

Please anyone can reply on the issue of 412 conditional request failed.



On Thu, Jan 18, 2024 at 6:49 PM satyaprakash ch <
chiramchetty.satyaprak...@gmail.com> wrote:

> Hi
>
> There were two PUBLISH requests were sent by the Kamailio Proxy server to
> the Kamailio Presence server.
>
> Processing of the first PUBLISH request resulted in the presentity table
> being updated - the "etag" changed from "a.1705053846.16611.21.1" to
> "a.1705053846.16613.23.2"
>
> Processing of the second PUBLISH failed because it also attempted to
> update the etag in the presentity table but at this point in time there was
> no match to the original etag value in the database
>
>
> Logs in kamailio.log:
> --
> ps_db_update_presentity(): No E-Tag match a.1705053846.16611.21.1
> DEBUG: tm [t_reply.c:637]: _reply_light(): reply sent out -
> buf=0x7f81e5ca5fc0: SIP/2.0 412 Conditio... shmem=0x7f81de6da068:
> SIP/2.0 412 Conditional request failed.
>
>
>
> Will any one suggest how to resolve the issue of 412 conditional request
> failed.
>
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[SR-Users] Re: How to get media IP

2024-01-24 Thread Alex Balashov via sr-users
Hi,

A complication of your request is that SDP can contain an arbitrary number of 
media sections for different streams. That's not needless pedantry, but rather 
to say that this makes it hard to create a straightforward, scalar 
pseudovariable around a concept like "the media IP". Besides that, Kamailio 
message pseudovariables are relatively high-level and mostly confined to SIP 
attributes. 

Nevertheless, your best bet, in terms of effort-to-reward ratio, is probably:

https://kamailio.org/docs/modules/5.7.x/modules/sdpops.html#sdpops.f.sdp_get_line_startswith

-- Alex

> On 24 Jan 2024, at 04:33, mm e via sr-users  
> wrote:
> 
> When receiving a call I get the following message:
> 
> INVITE sip:123456789@192.168.1.118:29363 SIP/2.0
> Via: SIP/2.0/UDP 
> 192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport
> Max-Forwards: 70
> Contact: 
> To: "123456789"
> From: "96341";tag=af3c7c31
> Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
> INFO
> Content-Type: application/sdp
> User-Agent: eyeBeam release 1011d stamp 40820
> Content-Length: 339
> 
> v=0
> o=- 3 2 IN IP4 192.168.1.118
> s=CounterPath eyeBeam 1.5
> c=IN IP4 192.168.1.118
> t=0 0
> m=audio 32170 RTP/AVP 0 8 18 101
> a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170
> a=fmtp:18 annexb=no
> a=fmtp:101 0-15
> a=rtpmap:18 G729/8000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
> 
> How to print the media IP in SDP separately in the log? I didn't find the 
> Pseudo-Variables..can you help me
> __
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-- 
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800

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[SR-Users] Re: Is the Contact header required for the dialog module?

2024-01-24 Thread Alex Balashov via sr-users
Hi,

Contact is mandatory in INVITEs. Per RFC 3261 § 8.1.1.8 ("Contact"):

   The Contact header field MUST be present and contain exactly one SIP
   or SIPS URI in any request that can result in the establishment of a
   dialog.  For the methods defined in this specification, that includes
   only the INVITE request.

-- Alex

> On 24 Jan 2024, at 05:42, Unai Rodriguez via sr-users 
>  wrote:
> 
> Dear List,
> 
> I’m trying to set up the dialog module and I keep getting the error "bad sip 
> message or missing Contact hdr”:
> 
> ---
> Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 
> 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog 
> [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing Contact 
> hdr
> Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 
> 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog 
> [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the 
> dialog
> ---
> 
> This is for MRCP traffic and the SIP INVITEs do not contain the Contact 
> header. They look like this:
> 
> 
> INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
> Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
> Max-Forwards: 70
> From: ;tag=134a3eKD2a8Ua
> To: 
> Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
> CSeq: 78491159 INVITE
> User-Agent: google_stt
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
> REFER, UPDATE
> Supported: timer, 100rel
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 329
> 
> v=0
> o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
> s=-
> c=IN IP4 10.10.10.10
> t=0 0
> m=application 9 TCP/MRCPv2 1
> a=setup:active
> a=connection:new
> a=resource:speechrecog
> a=cmid:1
> m=audio 12034 RTP/AVP 0 8 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 L16/8000
> a=sendonly
> a=mid:1
> 
> 
> According to the RFC, the Contact header is not mandatory 
> (https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t 
> have an easy way of adding the Contact header to the INVITEs. Is there a way 
> to configure Kamailio’s dialog module to do without the Contact header? Any 
> pointers appreciated.
> 
> Thank you so much
> 
> With best wishes, 
> Unai Rodriguez
> __
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-- 
Alex Balashov
Principal Consultant
Evariste Systems LLC
Web: https://evaristesys.com
Tel: +1-706-510-6800

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[SR-Users] Re: Is the Contact header required for the dialog module?

2024-01-24 Thread Who AmI via sr-users
Hi Unai,

The CONTACT is used as the R-URI in subsequent replies so upstream
providers parse this from my experience.

You can build one yourself in Kamailio if needed using the textops module -
https://kamailio.org/docs/modules/5.7.x/modules/textops.html#textops.f.append_hf

Rough examples:
Native:
insert_hf("Contact: $fu\r\n");

Kemi Python:
KSR.textopsx.append_hf_value("Contact", "")

Thanks,

John.

On Wed, 24 Jan 2024 at 11:15, Unai Rodriguez via sr-users <
sr-users@lists.kamailio.org> wrote:

> Dear List,
>
> I’m trying to set up the dialog module and I keep getting the error "bad
> sip message or missing Contact hdr”:
>
> ---
> Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1
> 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog
> [dlg_handlers.c:219]: populate_leg_info(): bad sip message or missing
> Contact hdr
> Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1
> 78451036 INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog
> [dlg_handlers.c:952]: dlg_new_dialog(): could not add further info to the
> dialog
> ---
>
> This is for MRCP traffic and the SIP INVITEs do not contain the Contact
> header. They look like this:
>
> 
> INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
> Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
> Max-Forwards: 70
> From: ;tag=134a3eKD2a8Ua
> To: 
> Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
> CSeq: 78491159 INVITE
> User-Agent: google_stt
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE
> Supported: timer, 100rel
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 329
>
> v=0
> o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
> s=-
> c=IN IP4 10.10.10.10
> t=0 0
> m=application 9 TCP/MRCPv2 1
> a=setup:active
> a=connection:new
> a=resource:speechrecog
> a=cmid:1
> m=audio 12034 RTP/AVP 0 8 96
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:96 L16/8000
> a=sendonly
> a=mid:1
> 
>
> According to the RFC, the Contact header is not mandatory (
> https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t
> have an easy way of adding the Contact header to the INVITEs. Is there a
> way to configure Kamailio’s dialog module to do without the Contact header?
> Any pointers appreciated.
>
> Thank you so much
>
> With best wishes,
> Unai Rodriguez
> __
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> the sender!
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[SR-Users] Is the Contact header required for the dialog module?

2024-01-24 Thread Unai Rodriguez via sr-users
Dear List,

I’m trying to set up the dialog module and I keep getting the error "bad sip 
message or missing Contact hdr”:

---
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 
INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:219]: 
populate_leg_info(): bad sip message or missing Contact hdr
Jan 23 11:34:17 server.example.com kamailio[2495]: 16(2511) ERROR: {1 78451036 
INVITE 2ef1d082-3486-123d-0abc-02c8e9c6c24e} dialog [dlg_handlers.c:952]: 
dlg_new_dialog(): could not add further info to the dialog
---

This is for MRCP traffic and the SIP INVITEs do not contain the Contact header. 
They look like this:


INVITE sip:some.mrcp.server.example.com:8060 SIP/2.0
Via: SIP/2.0/TCP 10.10.10.10:5099;branch=z9hG4bK5Kpey3r13SBNQ
Max-Forwards: 70
From: ;tag=134a3eKD2a8Ua
To: 
Call-ID: 051df0ff-3541-123d-f090-02f373ff014c
CSeq: 78491159 INVITE
User-Agent: google_stt
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, 
REFER, UPDATE
Supported: timer, 100rel
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 329

v=0
o=FreeSWITCH 2474358185109188492 5282339568667506935 IN IP4 10.10.10.10
s=-
c=IN IP4 10.10.10.10
t=0 0
m=application 9 TCP/MRCPv2 1
a=setup:active
a=connection:new
a=resource:speechrecog
a=cmid:1
m=audio 12034 RTP/AVP 0 8 96
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 L16/8000
a=sendonly
a=mid:1


According to the RFC, the Contact header is not mandatory 
(https://datatracker.ietf.org/doc/html/rfc3261#section-8.1.1) and I don’t have 
an easy way of adding the Contact header to the INVITEs. Is there a way to 
configure Kamailio’s dialog module to do without the Contact header? Any 
pointers appreciated.

Thank you so much

With best wishes,
Unai Rodriguez
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[SR-Users] Re: Roadmap to next major Kamailio release series v5.8.x

2024-01-24 Thread Daniel-Constantin Mierla via sr-users
Hello,

would it be possible to get the skeleton in the git repo or as PR for
the new module by end of the month? It is still a week and completing
missing parts can be done during the testing phase.

Cheers,
Daniel

On 23.01.24 10:13, Henning Westerholt wrote:
> Hello Daniel,
>
> we are working on a new module right now (related to special logging for high 
> CPS setups), which would be nice to integrate if possible. Its isolated, so 
> would not touch any other module and/or the core. There is also another core 
> refactoring in progress (to replace the atomic operations with the standard 
> Linux kernel support for better support of non-Intel/AMD architectures), but 
> this probably needs to wait to the next release cycle then.
>
> Best regards,
>
> Henning Westerholt
>
> -- 
> Henning Westerholt – https://skalatan.de/blog/
> Kamailio services – https://gilawa.com
>
>> -Original Message-
>> From: Daniel-Constantin Mierla 
>> Sent: Montag, 15. Januar 2024 11:33
>> To: Henning Westerholt ; Kamailio (SER) - Devel Mailing List
>> 
>> Cc: busin...@lists.kamailio.org; Kamailio (SER) - Users Mailing List > us...@lists.kamailio.org>
>> Subject: Re: [Kamailio-Business] Roadmap to next major Kamailio release
>> series v5.8.x
>>
>> Hello,
>>
>> first week of February is a semester school break in many regions, if someone
>> announces that he has ongoing work to a new module/feature, then freezing
>> can be postponed.
>>
>> But if not, I would rather do it earlier than waiting for nothing, because
>> starting later could end up in getting closer the Catholic Easter, which is 
>> earlier
>> this year, by end of March, a week before another holiday starts. Considering
>> the constraints on late March and April, if not out by mid of March or so, 
>> next
>> time frame will be end of April or during May.
>>
>> Cheers,
>> Daniel
>>
>> On 11.01.24 17:12, Henning Westerholt wrote:
>>> Hello,
>>>
>>> early February as a freezing date sounds good to me, just to have one full
>> month after the holiday season/vacation period to allow other people to 
>> finish
>> eventual contributions.
>>> Cheers,
>>>
>>> Henning
>>>
>>> --
>>> Henning Westerholt - https://skalatan.de/blog/ Kamailio services -
>>> https://gilawa.com
>>>
 -Original Message-
 From: Daniel-Constantin Mierla via business
 
 Sent: Mittwoch, 10. Januar 2024 10:11
 To: Kamailio (SER) - Users Mailing List
 ; Kamailio
 (SER) - Devel Mailing List ;
 busin...@lists.kamailio.org
 Subject: [Kamailio-Business] Roadmap to next major Kamailio release
 series v5.8.x

 Hello,

 discussed a bit during the online Kamailio devel meeting, it is time
 to set the milestones towards the next major Kamailio release series 
 v5.8.x.

 If no other suggestions that suit more developers, I would propose to
 freeze by end of this month or early February, then test for about 4
 weeks as usual and release by end of February or during March.

 If anyone wants to add new features/modules, they have to be
 published till freezing date, either pushed in the git repository or 
 proposed
>> as pull request.
 Cheers,
 Daniel

 --
 Daniel-Constantin Mierla (@ asipto.com) twitter.com/miconda --
 linkedin.com/in/miconda Kamailio Consultancy, Training and
 Development Services -- asipto.com Kamailio Advanced Training,
 February 20-22, 2024 -- asipto.com Kamailio World Conference, April
 18-19, 2024, Berlin -- kamailioworld.com

 ___
 Kamailio (OpenSER) - business mailing list --
 busin...@lists.kamailio.org To unsubscribe send an email to
 business-le...@lists.kamailio.org
 http://lists.openser-project.org/cgi-
 bin/mailman/listinfo%(cgiext)s/%(_internal_name)s
>> --
>> Daniel-Constantin Mierla (@ asipto.com)
>> twitter.com/miconda -- linkedin.com/in/miconda Kamailio Consultancy,
>> Training and Development Services -- asipto.com Kamailio Advanced Training,
>> February 20-22, 2024 -- asipto.com Kamailio World Conference, April 18-19,
>> 2024, Berlin -- kamailioworld.com

-- 
Daniel-Constantin Mierla (@ asipto.com)
twitter.com/miconda -- linkedin.com/in/miconda
Kamailio Consultancy, Training and Development Services -- asipto.com
Kamailio Advanced Training, February 20-22, 2024 -- asipto.com
Kamailio World Conference, April 18-19, 2024, Berlin -- kamailioworld.com

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[SR-Users] Re: Kamailio 5.6 (and 5.7) core dumping.

2024-01-24 Thread Dr . Barabás Péter via sr-users
Hi Mattis,

I have similar cases:
https://github.com/kamailio/kamailio/issues/3522

Question: is $uac_req(evroute) set to 1 and do you handle uac:reply event route?
In my case crash only occurs when these are true. If I switch evroute off, no 
crash occurs.

Péter Barabás



From: Mattis Lind via sr-users 
Date: Wednesday, 2024. January 24. 9:46
To: Kamailio (SER) - Users Mailing List 
Cc: Mattis Lind 
Subject: [SR-Users] Kamailio 5.6 (and 5.7) core dumping.
Hello Kamailio list!

We have a scenario that makes use of the UAC-module to send SIP MESSAGE and 
then in some cases the Kamailio process core dumps after some time after 
processing messages. I have been able to gather a core dump which shows this 
backtrace appended below. We are using Kamailio 5.6 retrieved from the kamailio 
repository: http://deb.kamailio.org/kamailio56. We are running Kamailio in a 
Docker container which runs on "5.10.0-25-cloud-amd64 #1 SMP Debian 5.10.191-1 
(2023-08-16) x86_64 GNU/Linux"

We have previously tried to use Kamailio 5.7 but it gave the same type of 
crashes.

We have been using the uac module a lot but it is just in this scenario we get 
a core dump. We have some kind of relation to a specific scenario but it can 
take from several seconds from the last SIP message of this scenario up to 50 
minutes until the crash occurs. To me this sounds like some kind of cleanup 
that is not handled properly. The back trace indicates that free of shared 
memory could be the issue, but I don't know the code unfortunately.

The last things we see in the log file is:

2024-01-23T13:18:08.828+01:00 Jan 23 12:18:08  /usr/sbin/kamailio[4789]: INFO: 

[SR-Users] How to get media IP

2024-01-24 Thread mm e via sr-users
When receiving a call I get the following message:

INVITE sip:123456789@192.168.1.118:29363 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.118:65178;branch=z9hG4bK-d87543-4a4a6b6c49541b2c-1--d87543-;rport
Max-Forwards: 70
Contact: 
To: "123456789"
From: "96341";tag=af3c7c31
Call-ID: OTFhN2M2YmRmNmU2N2I1ZmQxNmM4ODg4YzRiZmQyNTc.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, 
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1011d stamp 40820
Content-Length: 339

v=0
o=- 3 2 IN IP4 192.168.1.118
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.118
t=0 0
m=audio 32170 RTP/AVP 0 8 18 101
a=alt:1 1 : +YhwgPgy ayLEx4wo 192.168.1.118 32170
a=fmtp:18 annexb=no
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

How to print the media IP in SDP separately in the log? I didn't find the 
Pseudo-Variables..can you help me
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