[asterisk-users] Asterisk 13.1.0/PJSIP peer IP address issue
Scott Griepentrog
sgriepentrog at digium.com
Fri Jan 9 08:01:44 CST 2015
To sort out RTP problems, I would recommend:
1) on all endpoints use codec of allow=!all,ulaw -- this is or should be
supported by all endpoints and eliminates any issues of mismatch,
translation, etc., and can be adjusted later once everything is working
2) add an Echo() application to your dialplan so you can call it and check
RTP to and from Asterisk
3) start with direct_media=no to run all the RTP through Asterisk first
4) packet capture at/on the asterisk server, as well as at endpoints if
need be, to identfy if and where RTP streams are being sent and received.
The goal being to get two way audio calls up through Asterisk, and then
change one thing at a time towards your desired configuration and retest.
On Thu, Jan 8, 2015 at 7:03 PM, Sonny Rajagopalan <
sonny.rajagopalan at gmail.com> wrote:
> Well, I thought it worked, but it actually doesn't--I am able to get the
> caller pick up the phone, but for some reason, I cannot hear anything on
> either side no matter who does the calling. Again, my two SIP phones are on
> the local 192.168.1.0/24 network (do not go over the Internet) and the
> Asterisk server is located in the same network (not accessed over the
> Internet). Any help is appreciated.
>
> Does the fact that Asterisk is running on a VirtualBox VM on the same
> machine as one of the SIP phones matter? I am able to access the ARI REST
> interface of the Asterisk server quite fine on the host machine.
>
> I suspect it has to do with RTP not being set up, but all the codec
> support is there. Here's a log for the SIP request from 192.168.1.50:
>
> <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 --->
> INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 146.115.163.234:64009
> ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
> Max-Forwards: 70
> Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP>
> To: <sip:6002 at 192.168.1.139;transport=UDP>
> From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b
> Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO,
> SUBSCRIBE
> Content-Type: application/sdp
> Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
> User-Agent: Z 3.3.21933 r21903
>
> Authorization: Digest
> username="demo-alice",realm="asterisk",nonce="[removed]",uri="
> sip:6002 at 192.168.1.139
> ;transport=UDP",response="[removed]",cnonce="[removed]",nc=00000001,qop=auth,algorithm=md5,opaque="[removed]"
>
> Allow-Events: presence, kpml
> Content-Length: 245
>
>
> v=0
> o=Z 0 0 IN IP4 146.115.163.234
> s=Z
> c=IN IP4 146.115.163.234
> t=0 0
> m=audio 8000 RTP/AVP 0 3 110 8 98 101
> a=rtpmap:110 speex/8000
> a=rtpmap:98 iLBC/8000
> a=fmtp:98 mode=20
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
>
>
> <--- Transmitting SIP response (319 bytes) to UDP:192.168.1.50:64009 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 146.115.163.234:64009
> ;rport=64009;received=192.168.1.50;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z-
> Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE.
> From: <sip:demo-alice at 192.168.1.139>;tag=b661670b
> To: <sip:6002 at 192.168.1.139>
> CSeq: 2 INVITE
> Content-Length: 0
>
> Any help is appreciated. A topology is shown below in ASCII.
>
>
> < ( Big bad Internet ) >
>
> GW/NAPT/Router
> |
> ----------------------------------------------------------
> / \
>
> | |
> Host A Host B
> -----------------
> -----------------
> | Alice | | Bob
> |
> | 192.168.1.50 | |
> 192.168.1.149 |
> |---------------|
> |---------------|
> | Asterisk sr |
> | (VM) |
> | 192.168.1.239 |
> |---------------|
>
> On Thu, Jan 8, 2015 at 2:32 PM, Sonny Rajagopalan <
> sonny.rajagopalan at gmail.com> wrote:
>
>> Thank you for your note, Scott.
>>
>> I set rewrite_contact=yes for both contacts, and I also had to do
>> remove_existing=yes because I had to remove the existing contact
>> information (max_contacts = 1 was preventing new contact information)
>> using pjsip qualify demo-alice etc., after which the right IP addresses
>> showed in pjsip show endpoints. Anyway, it works as expected now, I
>> think. My pjsip.conf is now
>>
>> [transport-udp]
>> type=transport
>> protocol=udp
>> bind=0.0.0.0
>> local_net=192.168.1.0/24
>> ;Templates for the necessary config sections
>>
>> [endpoint_internal](!)
>> type=endpoint
>> context=from-internal
>> disallow=all
>> allow=ulaw
>>
>> [auth_userpass](!)
>> type=auth
>> auth_type=userpass
>>
>> [aor_dynamic](!)
>> type=aor
>> max_contacts=1
>> remove_existing=yes
>> ;Definitions for our phones, using the templates above
>>
>> [demo-alice](endpoint_internal)
>> auth=demo-alice
>> aors=demo-alice
>> mailboxes=box_a
>> rewrite_contact=yes
>> [demo-alice](auth_userpass)
>> password=demo-alice ; put a strong, unique password here instead
>> username=demo-alice
>>
>> [demo-alice](aor_dynamic)
>>
>> [demo-bob](endpoint_internal)
>> auth=demo-bob
>> aors=demo-bob
>> mailboxes=box_b
>> rewrite_contact=yes
>> [demo-bob](auth_userpass)
>> password=demo-bob ; put a strong, unique password here instead
>> username=demo-bob
>>
>> [demo-bob](aor_dynamic)
>>
>>
>> Thank you for your help!
>>
>> On Thu, Jan 8, 2015 at 11:48 AM, Scott Griepentrog <
>> sgriepentrog at digium.com> wrote:
>>
>>> It would appear that you have the Asterisk server on a public IP
>>> address, your two endpoints are behind a NAT, and you have rewrite_contact
>>> enabled in pjsip.conf.
>>>
>>> In which case, what you are seeing is correct. In order to be able to
>>> send a call to an extension where it is behind NAT, Asterisk must update
>>> the contact to have the current IP and port that the phone registered via
>>> (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state
>>> for).
>>>
>>> On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan <
>>> sonny.rajagopalan at gmail.com> wrote:
>>>
>>>> I am following the instructions in
>>>> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and
>>>> I am trying to make a call from extension Alice (6001) to extension for Bob
>>>> (6002). When I make the call, I can hear the ringing on Alice's phone
>>>> (caller), but Bob's phone (callee) doesn't ring, or show a call coming in
>>>> from Alice. My setup and environment is as follows: Alice, Bob and Asterisk
>>>> all in the same 192.168.1.0/24 network, and they are able to register
>>>> to the Asterisk server running 13.1.0/PJSIP. The rest of the configuration
>>>> is the same as the aforementioned wiki page, but is shown here for clarity:
>>>>
>>>> root at asterisk13FFP:/var/log/asterisk# more
>>>> /etc/asterisk/extensions.conf
>>>> [from-internal]
>>>> exten=>6001,1,Dial(PJSIP/demo-alice)
>>>> exten=>6002,1,Dial(PJSIP/demo-bob)
>>>> exten=>6003,1,Answer()
>>>> same =>6003,n,Playback(hello-world)
>>>> same =>6003,n,Hangup()
>>>>
>>>>
>>>> What I do observe is that I when I request the output of pjsip show
>>>> endpoints, I get Contact information for the two SIP peers that have
>>>> registered different from their actual IP addresses. I suspect this has
>>>> something to do with their calls being routed elsewhere. If my assumption
>>>> is correct--how do I fix this? Alice should be at 192.168.1.50 and Bob
>>>> should be at 192.168.1.149, instead, they (both) show IP address
>>>> 146.115.163.234. Any help is deeply appreciated. Thanks.
>>>>
>>>> asterisk13FFP*CLI> pjsip show endpoints
>>>>
>>>> Endpoint: <Endpoint/CID.....................................>
>>>> <State.....> <Channels.>
>>>> I/OAuth:
>>>> <AuthId/UserName...........................................................>
>>>> Aor: <Aor............................................>
>>>> <MaxContact>
>>>> Contact: <Aor/ContactUri...............................>
>>>> <Status....> <RTT(ms)..>
>>>> Transport: <TransportId........> <Type> <cos> <tos>
>>>> <BindAddress..................>
>>>> Identify:
>>>> <Identify/Endpoint.........................................................>
>>>> Match: <ip/cidr.........................>
>>>> Channel: <ChannelId......................................>
>>>> <State.....> <Time(sec)>
>>>> Exten: <DialedExten...........> CLCID:
>>>> <ConnectedLineCID.......>
>>>>
>>>> =========================================================================================
>>>>
>>>> Endpoint: demo-alice
>>>> Unavailable 0 of inf
>>>> InAuth: demo-alice/demo-alice
>>>> Aor: demo-alice 1
>>>> Contact: demo-alice/sip:demo-alice@*146.115.163.234*:38519
>>>> Unknown nan
>>>>
>>>> Endpoint: demo-bob Not in
>>>> use 0 of inf
>>>> InAuth: demo-bob/demo-bob
>>>> Aor: demo-bob 1
>>>> Contact: demo-bob/sip:demo-bob@*146.115.163.234*:38321;tra
>>>> Unknown nan
>>>>
>>>>
>>>> --
>>>> _____________________________________________________________________
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>>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>
>>>
>>> --
>>> [image: Digium logo]
>>> Scott Griepentrog
>>> Digium, Inc · Software Developer
>>> 445 Jan Davis Drive NW · Huntsville, AL 35806 · US
>>> direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
>>> Check us out at: http://digium.com · http://asterisk.org
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>> http://www.asterisk.org/hello
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
> http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 256 580 6090
Check us out at: http://digium.com · http://asterisk.org
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