[asterisk-users] Asterisk behind NAT Early Media Video

Wed Apr 11 01:54:57 CDT 2018

I think I found the root cause. The H264 Early Media video is received
successfully on the Asterisk Server. It also seems to get processed. But
it's send to the private IP of the receipent SIP phone.

For clarification:
178.82.XX.XX is my Public IP of my Internet access. Both phones use this as
Public IP via standard Source NAT.
159.89.XX.XX is the IP of the Asterisk Server. For this test I used a
Server without Destination NAT. So the eth0 interface has this IP.

Packet capture:
No.     Time                          Source
Destination           Protocol Length Info
    141 2018-04-11 06:40:03.306561    178.82.XX.XX          159.89.XX.XX
       H264     64     PT=H264, SSRC=0x3CB1E12D, Seq=19561, Time=319121408
SPS

Frame 141: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: JuniperN_34:67:f0 (40:a6:77:34:67:f0), Dst:
da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7)
Internet Protocol Version 4, Src: 178.82.169.0, Dst: 159.89.104.193
User Datagram Protocol, Src Port: 5006, Dst Port: 13182
Real-Time Transport Protocol
H.264

No.     Time                          Source
Destination           Protocol Length Info
    142 2018-04-11 06:40:03.306682    159.89.XX.XX
192.168.XX.XX         H264     64     PT=H264, SSRC=0x5EE97C55, Seq=30572,
Time=319121408 SPS

Frame 142: 64 bytes on wire (512 bits), 64 bytes captured (512 bits)
Ethernet II, Src: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7), Dst:
IETF-VRRP-VRID_6e (00:00:5e:00:01:6e)
Internet Protocol Version 4, Src: 159.89.104.193, Dst: 192.168.1.185
User Datagram Protocol, Src Port: 10298, Dst Port: 5022
Real-Time Transport Protocol
H.264

PJSIP.conf:
[7004]
type = endpoint
context = internal
rewrite_contact = yes
direct_media = no
rtp_symmetric = yes
;force_rport = yes
disallow = all
allow = g722, alaw, ulaw, gsm, ilbc, h264
aors = 7004
auth = auth7004

[7004]
type = aor
max_contacts = 2

[auth7004]
type=auth
auth_type=userpass
password=1234
username=7004

extensions.conf:
[internal]
exten => _700X,1,Dial(PJSIP/${EXTEN})

2018-04-10 16:43 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>:

> I just noticed, the calling device isn't even sending the early media
> video stream. It just sends an early media audio stream. Is there propably
> a change in the signaling needed?
>
> (On another P2P SIP Server the early media video works.)
>
> 2018-04-10 12:29 GMT+02:00 Benjamin Marty <benjamin.marty at gmail.com>:
>
>> Hi Florian
>>
>> I already have the external_media_address set in the PJSIP setup. Also
>> the external_signaling_address is set to the Public IP. If I make a call
>> from an Early Media (video&audio) capable device to an Early Media capable
>> device (also video&audio) the Early Media audio works perfectly. But no
>> video. If I sniff with wireshark on the recipent device I just see G711
>> (audio) RTP traffic. The h264 RTP traffic is missing before I accept the
>> call. After accepting the call the h264 RTP traffic comes through.
>>
>> The 183 SIP protocoll comes through. Even Asterisk is noticing it:
>> -- PJSIP/6002-00000013 is making progress passing it to
>> PJSIP/6001-00000012
>>
>> I tried both Asterisk 15 with pjsip.conf configuration and Asterisk 13
>> with sip.conf (chan_sip). In both cases I just put the both case
>> AST_FRAME_VIDEO: statements before the two voice cases, like in your diff
>> and recompiled/reinstalled.
>>
>> Regards
>>
>> Benjamin
>>
>>
>>
>> 2018-04-10 9:37 GMT+02:00 Floimair Florian <f.floimair at commend.com>:
>>
>>> Hi Benjamin!
>>>
>>> You're obviously using a similar scenario that I have in place for
>>> testing.
>>> I initially had issues with early media (not only video also audio) as
>>> well in that scenario. What I had to do was to additionally set
>>>
>>> external_media_address=<your external IP>
>>>
>>> in pjsip.conf
>>>
>>> Also, as I wrote the patch for early-media video I'd be interested in
>>> any feedback from it.
>>>
>>>
>>>
>>>
>>> With best regards
>>>
>>> Florian Floimair
>>> Innovation - Software-Development -  VoIP & DevOps
>>>
>>> COMMEND INTERNATIONAL GMBH
>>> A-5020 Salzburg, Saalachstraße 51
>>> Tel: +43-662-85 62 25
>>> Fax: +43-662-85 62 26
>>> http://www.commend.com
>>>
>>> Security and Communication by Commend
>>>
>>> FN 178618z | LG Salzburg
>>>
>>> -----Ursprüngliche Nachricht-----
>>> Von: asterisk-users-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at lists.digium.com] Im Auftrag von Joshua Colp
>>> Gesendet: Montag, 9. April 2018 18:15
>>> An: asterisk-users at lists.digium.com
>>> Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video
>>>
>>> On Mon, Apr 9, 2018, at 1:05 PM, Benjamin Marty wrote:
>>> > wohoo, so if I unterstand it correctly with that patch early media
>>> > video works over the Asterisk server? In other words the Asterisk
>>> > server get's able to (process/)forward the early media video stream
>>> with that patch?
>>>
>>> The patch forwards video while in an early media state before the call
>>> is answered and bridged, yes.
>>>
>>> --
>>> Joshua Colp
>>> Digium, Inc. | Senior Software Developer
>>> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at:
>>> https://linkprotect.cudasvc.com/url?a=https%3a%2f%2fwww.digi
>>> um.com&c=E,1,fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnE
>>> pbgeqpqYbk39h-m_lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduN
>>> naBqVCDk,&typo=1 & www.asterisk.org
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.co
>>> m/url?a=http%3a%2f%2fwww.api-digital.com&c=E,1,XToemLgPy6NQV
>>> yb_dF1q0qXSk-3YylF6rmIrWQvPhspxagnF5G63VHCU2nB67YHjZewMQU1rU
>>> CME4JBQMFPmNOCpc6ESOin_3Al6kti-lRo,&typo=1 --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    https://linkprotect.cudasvc.com/url?a=http%3a%2f%2flists.di
>>> gium.com%2fmailman%2flistinfo%2fasterisk-users&c=E,1,6VfJH-y
>>> sYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmo
>>> l-9H658j2bZr9JmLmb9WCM5OXKTsb_DsBIYKACtBorWRSU6-q1FjJkrbc&typo=1
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>>
>>> Check out the new Asterisk community forum at:
>>> https://community.asterisk.org/
>>>
>>> New to Asterisk? Start here:
>>>       https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>
>>
>
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