Re: [asterisk-users] SSRC =0x0 in RTP

2017-11-15 Thread Joshua Colp
On Wed, Nov 15, 2017, at 01:37 PM, Harel Cohen wrote:
> >
> > Hi Joshua,
> 
> > Thank you for looking into this.
> 
> > Their response IS based on traces I've sent them. Attached is such trace
> in text format (server IP has been changed to 111.111.111.111). Some
> repeating RTP packets has been truncated.
> You can see that after the 200 OK SSRC is sent from the server to the
> phone
> as '0x0'. The same has happened with G729 codec.
> 
> > Let me know if you need the full trace or anything else from my side.
> 
> > I should also mention that this is Asterisk version 1.8.12.1

I'm sorry but this version is old enough that what I currently know is
far past it. It may have been possible in that old version for the SSRC
to be as you state. In recent stuff it doesn't seem to be possible.

-- 
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

-- 
_
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Re: [asterisk-users] SSRC =0x0 in RTP

2017-11-15 Thread Harel Cohen
>
> Hi Joshua,

> Thank you for looking into this.

> Their response IS based on traces I've sent them. Attached is such trace
in text format (server IP has been changed to 111.111.111.111). Some
repeating RTP packets has been truncated.
You can see that after the 200 OK SSRC is sent from the server to the phone
as '0x0'. The same has happened with G729 codec.

> Let me know if you need the full trace or anything else from my side.

> I should also mention that this is Asterisk version 1.8.12.1

> Thank you

> Harel

> --

>
On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote:

> > Hello,

> > I have a problem where on an outgoing call a Grandstream phone

> > (GXP2130) closes the incoming voice stream about 1 second into the

> > call (the remote party hears the Grandstream, the Grandstream doesn't

> > hear thr remote party). I have verified with logs and traces that this

> > is not a NAT issue or any other network-related problem. All incoming

> > RTP packets arrive at the phone on the correct port etc. as declared in
the SDP.

> > I opened a ticket with Grandstream and they replied: "

> >

> > *the phone starts receiving RTP with SSRC =0x0 which is wrong".*

> >

> > Is this an Asterisk problem or the phones? Is this something that can

> > be fixed on the Asterisk side?

>
Asterisk would be sending the RTP to the Grandstream. I'd suggest getting a
packet capture using tcpdump or wireshark to confirm what they've said
though. I just looked at the code and I don't see a way that we'd ever have
the SSRC be 0.

>
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

>

>
No. Time   SourceDestination   Protocol 
Length Info
  1 0.00
1224   

Frame 1: 1224 bytes on wire (9792 bits), 1224 bytes captured (9792 bits)
This frame is marked as ignored

No. Time   SourceDestination   Protocol 
Length Info
  2 0.078267   195.191.156.50192.168.10.144SIP  631 
   Status: 401 Unauthorized | 

Frame 2: 631 bytes on wire (5048 bits), 631 bytes captured (5048 bits)
Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa 
(00:0b:82:a7:a0:fa)
802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0
Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144
User Datagram Protocol, Src Port: 55060, Dst Port: 21321
Session Initiation Protocol (401)

No. Time   SourceDestination   Protocol 
Length Info
  3 0.090271   192.168.10.144195.191.156.50SIP  431 
   Request: ACK sip:0035699143...@sip1.mayorcom.com:55060 | 

Frame 3: 431 bytes on wire (3448 bits), 431 bytes captured (3448 bits)
Ethernet II, Src: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa), Dst: 00:ff:6b:60:2c:4c 
(00:ff:6b:60:2c:4c)
Internet Protocol Version 4, Src: 192.168.10.144, Dst: 195.191.156.50
User Datagram Protocol, Src Port: 21321, Dst Port: 55060
Session Initiation Protocol (ACK)

No. Time   SourceDestination   Protocol 
Length Info
  4 0.120599   192.168.10.144195.191.156.50SIP/SDP  
1427   Request: INVITE sip:0035699143...@sip1.mayorcom.com:55060 | 

Frame 4: 1427 bytes on wire (11416 bits), 1427 bytes captured (11416 bits)
Ethernet II, Src: Grandstr_a7:a0:fa (00:0b:82:a7:a0:fa), Dst: 00:ff:6b:60:2c:4c 
(00:ff:6b:60:2c:4c)
Internet Protocol Version 4, Src: 192.168.10.144, Dst: 195.191.156.50
User Datagram Protocol, Src Port: 21321, Dst Port: 55060
Session Initiation Protocol (INVITE)

No. Time   SourceDestination   Protocol 
Length Info
  5 0.200309   195.191.156.50192.168.10.144SIP  576 
   Status: 100 Trying | 

Frame 5: 576 bytes on wire (4608 bits), 576 bytes captured (4608 bits)
Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa 
(00:0b:82:a7:a0:fa)
802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0
Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144
User Datagram Protocol, Src Port: 55060, Dst Port: 21321
Session Initiation Protocol (100)

No. Time   SourceDestination   Protocol 
Length Info
  6 1.742472   195.191.156.50192.168.10.144SIP/SDP  883 
   Status: 183 Session Progress | 

Frame 6: 883 bytes on wire (7064 bits), 883 bytes captured (7064 bits)
Ethernet II, Src: 00:ff:6b:60:2c:4c (00:ff:6b:60:2c:4c), Dst: Grandstr_a7:a0:fa 
(00:0b:82:a7:a0:fa)
802.1Q Virtual LAN, PRI: 0, DEI: 0, ID: 0
Internet Protocol Version 4, Src: 195.191.156.50, Dst: 192.168.10.144
User Datagram Protocol, Src Port: 55060, Dst Port: 21321
Session Initiation Protocol (183)

No. Time   SourceDestination   Protocol 
Length Info

Re: [asterisk-users] SSRC =0x0 in RTP

2017-11-14 Thread Joshua Colp
On Tue, Nov 14, 2017, at 09:14 AM, Harel Cohen wrote:
> Hello,
> I have a problem where on an outgoing call a Grandstream phone (GXP2130)
> closes the incoming voice stream about 1 second into the call (the remote
> party hears the Grandstream, the Grandstream doesn't hear thr remote
> party). I have verified with logs and traces that this is not a NAT issue
> or any other network-related problem. All incoming RTP packets arrive at
> the phone on the correct port etc. as declared in the SDP.
> I opened a ticket with Grandstream and they replied: "
> 
> *the phone starts receiving RTP with SSRC =0x0 which is wrong".*
> 
> Is this an Asterisk problem or the phones? Is this something that can be
> fixed on the Asterisk side?

Asterisk would be sending the RTP to the Grandstream. I'd suggest
getting a packet capture using tcpdump or wireshark to confirm what
they've said though. I just looked at the code and I don't see a way
that we'd ever have the SSRC be 0.

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

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[asterisk-users] SSRC =0x0 in RTP

2017-11-14 Thread Harel Cohen
Hello,
I have a problem where on an outgoing call a Grandstream phone (GXP2130)
closes the incoming voice stream about 1 second into the call (the remote
party hears the Grandstream, the Grandstream doesn't hear thr remote
party). I have verified with logs and traces that this is not a NAT issue
or any other network-related problem. All incoming RTP packets arrive at
the phone on the correct port etc. as declared in the SDP.
I opened a ticket with Grandstream and they replied: "

*the phone starts receiving RTP with SSRC =0x0 which is wrong".*

Is this an Asterisk problem or the phones? Is this something that can be
fixed on the Asterisk side?

Thank you,

Harel
-- 
_
-- 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