Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Fri, Apr 13, 2018, at 11:56 AM, Benjamin Marty wrote: > The current behaviour is that Earlymedia video isn't working when NAT's in > between are involved. The source/destination IP's are correct. So the > client is sending Early media video + Early media audio to the Asterisk > Server "in the cloud" and the Asterisk Server "in the cloud" is sending > both to the IP where the Client is located. But strangely just the Early > media audio is passing the NAT to the recipent. > > My guess is that the NAT traversal for Early media audio is fine, but the > one for Early media video not yet. Can you propably comprehend something in > that direction? Or can you guide me to the code part where Asterisk is > doing the Port change when a NAT is detected and the Client itself is > sending "fake" RTP Early media traffic to get a NAT Binding for incoming > RTP Early media traffic? The code is in res_rtp_asterisk[1]. It's not complex and despite the comment is not specific to video. Without logs showing where things are coming from and going I don't really have anything else to add. [1] https://github.com/asterisk/asterisk/blob/master/res/res_rtp_asterisk.c#L6140 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
The current behaviour is that Earlymedia video isn't working when NAT's in between are involved. The source/destination IP's are correct. So the client is sending Early media video + Early media audio to the Asterisk Server "in the cloud" and the Asterisk Server "in the cloud" is sending both to the IP where the Client is located. But strangely just the Early media audio is passing the NAT to the recipent. My guess is that the NAT traversal for Early media audio is fine, but the one for Early media video not yet. Can you propably comprehend something in that direction? Or can you guide me to the code part where Asterisk is doing the Port change when a NAT is detected and the Client itself is sending "fake" RTP Early media traffic to get a NAT Binding for incoming RTP Early media traffic? Benjamin 2018-04-11 11:50 GMT+02:00 Joshua Colp : > On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > > I added the bind_rtp_to_media_address=yes on all endpoints but still the > > same behaviour. The funny thing is that the G711 audio early media works > > and doesn't have that Private IP issue. I was also able to cross check > with > > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > > capture (PJSIP): > > As I stated previously in order for media to go to the source IP address > and port, media has to be received from the endpoint. If this doesn't > happen then you'll see exactly this behavior - we'll send to the IP address > and port they told us. There's nothing that Asterisk itself can do in that > instance, the endpoint has to send media or place the correct IP address > and port in the messages. > > Was any media received from it? > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Wed, Apr 11, 2018, at 4:33 AM, Benjamin Marty wrote: > I added the bind_rtp_to_media_address=yes on all endpoints but still the > same behaviour. The funny thing is that the G711 audio early media works > and doesn't have that Private IP issue. I was also able to cross check with > chan_sip on Asterisk 15, exactly the same wrong behaviour. See following > capture (PJSIP): As I stated previously in order for media to go to the source IP address and port, media has to be received from the endpoint. If this doesn't happen then you'll see exactly this behavior - we'll send to the IP address and port they told us. There's nothing that Asterisk itself can do in that instance, the endpoint has to send media or place the correct IP address and port in the messages. Was any media received from it? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
I added the bind_rtp_to_media_address=yes on all endpoints but still the same behaviour. The funny thing is that the G711 audio early media works and doesn't have that Private IP issue. I was also able to cross check with chan_sip on Asterisk 15, exactly the same wrong behaviour. See following capture (PJSIP): No. Time Source Destination Protocol Length Info 187 2018-04-11 07:19:56.735967159.89.XX.XX 192.168.1.185 H264 943PT=H264, SSRC=0x3A7AF929, Seq=27144, Time=1248011648 FU-A Frame 187: 943 bytes on wire (7544 bits), 943 bytes captured (7544 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.XX.XX, Dst: 192.168.1.185 User Datagram Protocol, Src Port: 11502, Dst Port: 5022 Real-Time Transport Protocol H.264 No. Time Source Destination Protocol Length Info 188 2018-04-11 07:19:56.735993159.89.XX.XX 192.168.1.185 H264 943PT=H264, SSRC=0x3A7AF929, Seq=27145, Time=1248011648, Mark FU-A End Frame 188: 943 bytes on wire (7544 bits), 943 bytes captured (7544 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.XX.XX, Dst: 192.168.1.185 User Datagram Protocol, Src Port: 11502, Dst Port: 5022 Real-Time Transport Protocol H.264 No. Time Source Destination Protocol Length Info 189 2018-04-11 07:19:56.738966178.82.XX.XX 159.89.XX.XXRTP 214PT=ITU-T G.711 PCMU, SSRC=0x2A1A1C31, Seq=1820, Time=1104983225 Frame 189: 214 bytes on wire (1712 bits), 214 bytes captured (1712 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.XX.XX, Dst: 159.89.XX.XX User Datagram Protocol, Src Port: 5020, Dst Port: 16130 Real-Time Transport Protocol No. Time Source Destination Protocol Length Info 190 2018-04-11 07:19:56.738975178.82.XX.XX 159.89.XX.XXRTP 214PT=ITU-T G.722, SSRC=0x49CD55FD, Seq=26679, Time=470333826 Frame 190: 214 bytes on wire (1712 bits), 214 bytes captured (1712 bits) Ethernet II, Src: JuniperN_4f:3f:f0 (40:a6:77:4f:3f:f0), Dst: da:81:42:3d:d0:e7 (da:81:42:3d:d0:e7) Internet Protocol Version 4, Src: 178.82.XX.XX, Dst: 159.89.XX.XX User Datagram Protocol, Src Port: 5004, Dst Port: 18280 Real-Time Transport Protocol 2018-04-11 9:11 GMT+02:00 Floimair Florian : > I did a quick check between what I have set and your settings below. > > > > You can try the following and see if it helps > > > > In your endpoint: > bind_rtp_to_media_address=yes > > > > > > > > > > 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 > > > > *Von:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *Im Auftrag von *Benjamin Marty > *Gesendet:* Mittwoch, 11. April 2018 08:55 > *An:* Asterisk Users Mailing List - Non-Commercial Discussion < > asterisk-users@lists.digium.com> > > *Betreff:* Re: [asterisk-users] Asterisk behind NAT Early Media Video > > > > 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 06:40:03.306561178.82.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 06:40:03.30
Re: [asterisk-users] Asterisk behind NAT Early Media Video
I did a quick check between what I have set and your settings below.
You can try the following and see if it helps
In your endpoint:
bind_rtp_to_media_address=yes
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
Von: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Benjamin Marty
Gesendet: Mittwoch, 11. April 2018 08:55
An: Asterisk Users Mailing List - Non-Commercial Discussion
Betreff: Re: [asterisk-users] Asterisk behind NAT Early Media Video
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 SourceDestination
Protocol Length Info
141 2018-04-11 06:40:03.306561178.82.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 SourceDestination
Protocol Length Info
142 2018-04-11 06:40:03.306682159.89.XX.XX192.168.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
mailto:benjamin.ma...@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
mailto:benjamin.ma...@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/6002-0013 is making progress passing it to PJSIP/6001-0012
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
mailto:f.floim...@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=
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<https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.commend.com&c=E,1,3-QFS79bl07XJ1At9-FN042YWg_pIhOoaMJ3B13IzEVsdUP_-SFZDUg5wBrnkEzQgB7TrZRQzaiO0i
Re: [asterisk-users] Asterisk behind NAT Early Media Video
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 06:40:03.306561178.82.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 06:40:03.306682159.89.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 :
> 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 :
>
>> 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/6002-0013 is making progress passing it to
>> PJSIP/6001-0012
>>
>> 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 :
>>
>>> 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=
>>>
>>> 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-boun...@lists.digium.com [mailto:
>>> asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp
>>> Gesendet: Montag, 9. April 2018 18:15
>>> An: asterisk-users@lists.digium.com
>>> Betreff: Re: [aster
Re: [asterisk-users] Asterisk behind NAT Early Media Video
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 : > 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/6002-0013 is making progress passing it to PJSIP/6001-0012 > > 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 : > >> 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= >> >> 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-boun...@lists.digium.com [mailto: >> asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp >> Gesendet: Montag, 9. April 2018 18:15 >> An: asterisk-users@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,XToemLgPy6NQ >> Vyb_dF1q0qXSk-3YylF6rmIrWQvPhspxagnF5G63VHCU2nB67YHjZewMQU1r >> UCME4JBQMFPmNOCpc6ESOin_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- >> ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jm >> ol-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 >> > > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
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/6002-0013 is making progress passing it to PJSIP/6001-0012 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 : > 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= > > 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-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] Im Auftrag von Joshua Colp > Gesendet: Montag, 9. April 2018 18:15 > An: asterisk-users@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.digium.com&c=E,1, > fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnEpbgeqpqYbk39h-m_ > lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduNnaBqVCDk,&typo=1 & > www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc. > com/url?a=http%3a%2f%2fwww.api-digital.com&c=E,1, > XToemLgPy6NQVyb_dF1q0qXSk-3YylF6rmIrWQvPhspxagnF5G63VHCU > 2nB67YHjZewMQU1rUCME4JBQMFPmNOCpc6ESOin_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. > digium.com%2fmailman%2flistinfo%2fasterisk-users&c= > E,1,6VfJH-ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmol- > 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 > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
I applied the patch to my Asterisk 13.20. But it seems that it still
doesn't forward the early media video stream. Do I need to put something
special into the extensions.conf? I basically just make a Dial. The calling
Client sends the 183 protocol.
[public]
exten => 6001,1,Dial(SIP/${EXTEN})
2018-04-09 18:14 GMT+02:00 Joshua Colp :
> 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: www.digium.com & www.asterisk.org
>
> --
> _
> -- 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
>
--
_
-- 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
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Re: [asterisk-users] Asterisk behind NAT Early Media Video
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= 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-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] Im Auftrag von Joshua Colp Gesendet: Montag, 9. April 2018 18:15 An: asterisk-users@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.digium.com&c=E,1,fYho2t3OGEPSC6ILhV9IAhfyqyv57q-c2eodmmoTlhRYCnEpbgeqpqYbk39h-m_lDWff7UIltd0zakv3XGb858ysVJbX0qeWGwdsbcgvduNnaBqVCDk,&typo=1 & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by https://linkprotect.cudasvc.com/url?a=http%3a%2f%2fwww.api-digital.com&c=E,1,XToemLgPy6NQVyb_dF1q0qXSk-3YylF6rmIrWQvPhspxagnF5G63VHCU2nB67YHjZewMQU1rUCME4JBQMFPmNOCpc6ESOin_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.digium.com%2fmailman%2flistinfo%2fasterisk-users&c=E,1,6VfJH-ysYuWrel9Apl4EqHb4_MpDTQHdQ3lJU3_Zojgbn4stUdMfchlswYSSwVO9jmol-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
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: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
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? 2018-04-09 17:57 GMT+02:00 Joshua Colp : > On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > > My understanding based on Wireshark analysis is that the signaling works > > (also the recipent phone is displaying the video frame before accepting > the > > call), also the calling phone send video (i see that also via Wireshark) > > but the recipent phone doesn't get any video from the Asterisk before the > > call. > > Ah yeah video, I forgot that it was a recent change to add support for > it[1]. It's not yet in any release. > > [1] https://gerrit.asterisk.org/#/c/8398/ > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 12:04 PM, Benjamin Marty wrote: > My understanding based on Wireshark analysis is that the signaling works > (also the recipent phone is displaying the video frame before accepting the > call), also the calling phone send video (i see that also via Wireshark) > but the recipent phone doesn't get any video from the Asterisk before the > call. Ah yeah video, I forgot that it was a recent change to add support for it[1]. It's not yet in any release. [1] https://gerrit.asterisk.org/#/c/8398/ -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
My understanding based on Wireshark analysis is that the signaling works (also the recipent phone is displaying the video frame before accepting the call), also the calling phone send video (i see that also via Wireshark) but the recipent phone doesn't get any video from the Asterisk before the call. 2018-04-09 17:02 GMT+02:00 Joshua Colp : > On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote: > > Yes, media is flowing through Asterisk because both client's are behind > > different NAT's. > > This doesn't answer the question of what is ACTUALLY happening in the > scenario you describe which is very important. > > > Do I need to do something special in the Call Flow? Or anything > additional > > to the pjsip.conf? > > The "rtp_symmetric" option as you've used causes Asterisk to send media to > the source of media, but it requires us to receive media. If we don't > receive it then we send media to where they've told us to send it, which as > I've mentioned can be wrong. > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 11:53 AM, Benjamin Marty wrote: > Yes, media is flowing through Asterisk because both client's are behind > different NAT's. This doesn't answer the question of what is ACTUALLY happening in the scenario you describe which is very important. > Do I need to do something special in the Call Flow? Or anything additional > to the pjsip.conf? The "rtp_symmetric" option as you've used causes Asterisk to send media to the source of media, but it requires us to receive media. If we don't receive it then we send media to where they've told us to send it, which as I've mentioned can be wrong. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
Yes, media is flowing through Asterisk because both client's are behind different NAT's. Do I need to do something special in the Call Flow? Or anything additional to the pjsip.conf? 2018-04-09 16:50 GMT+02:00 Joshua Colp : > On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > > Hello, > > > > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). > > > > Now I would like to get Early Media Video working between clients in > > different NATed networks. The 183 signalling goes trough perfectly, but > > asterisk doesn't forward the Early Media RTP stream from the caller to > the > > recipent. > > You would need to examine things specifically and see where media is > flowing. Is the recipient behind NAT? If so then until we receive media > from them (wich may or may not occur with early media) we may not have the > correct target of media. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 > -- _ -- 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
Re: [asterisk-users] Asterisk behind NAT Early Media Video
On Mon, Apr 9, 2018, at 11:42 AM, Benjamin Marty wrote: > Hello, > > I have an Asterisk 15 with PJSIP behind NAT (Amazon EC2). > > Now I would like to get Early Media Video working between clients in > different NATed networks. The 183 signalling goes trough perfectly, but > asterisk doesn't forward the Early Media RTP stream from the caller to the > recipent. You would need to examine things specifically and see where media is flowing. Is the recipient behind NAT? If so then until we receive media from them (wich may or may not occur with early media) we may not have the correct target of media. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- 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
