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[Freeswitch-users] codec negotiation error

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brian at linuxpenguins...
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PostPosted: Wed Aug 03, 2022 3:28 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Hello,

I receiving an incoming call, with the G.729 PCMA codec. I try to bridge
this call to my VOIP provider. This generates the following SDP packet:

=== cut ===
v=0
o=FreeSWITCH 175841631 175841632 IN IP4 59.167.180.194
s=FreeSWITCH
c=IN IP4 59.167.180.194
t=0 0
m=audio 29188 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
=== cut ===

I think this is bad, as PCMA isn't listed in the codecs available.

How do I debug this?

I am assuming this might be the reason why I am not getting incoming
audio (does this make sense?)

Regards
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
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https://freeswitch.org/confluence
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brian at linuxpenguins...
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PostPosted: Wed Aug 03, 2022 4:03 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian May <brian@linuxpenguins.xyz> writes:

Quote:
I am assuming this might be the reason why I am not getting incoming
audio (does this make sense?)

Never mind, I found
https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp

Turned it on, works as expected, but still not getting incoming audio.

Arrgh!

So far everything I have looked at appears to be OK, no idea why I am
having audio problems.
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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brian at linuxpenguins...
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PostPosted: Wed Aug 03, 2022 4:11 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian May <brian@linuxpenguins.xyz> writes:

Quote:
Never mind, I found
https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp

Plus
https://freeswitch-users.freeswitch.narkive.com/jyFwqoeR/incompatible-destination-did-fs-not-offer-a-codec
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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piotr at dataandsignal...
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PostPosted: Sat Aug 06, 2022 10:28 am    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Quote:
m=audio 29188 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
=== cut ===
I think this is bad, as PCMA isn't listed in the codecs available.


Oops. You know what 8 in the m=audio line stands for Brian?
Yes, it is for PCMA.


Quote:
How do I debug this?


You can start from checking if these packets are sent at all, and if you get those packets at all.
Use wireshark or tcpdump.
Do the same on both machines, you will see if and where RTP is sent/recved.

If all RTP is sent but it cannot reach your FreeSWITCH, then check ports are allowed in the firewall. 


regards,


[img]https://ci3.googleusercontent.com/mail-sig/AIorK4wE8rSMg277YOGBrgEQayYWXH2G53bMgBu7uf-k-vU6x5SD1T6YWorVfbkDegPbnXcFyHwBODg[/img]



Piotr Gregor
Software Engineer


M: (+44) 07483 866 525     L: (+44) 01256 597 470     www: dataandsignal.com






















On Wed, Aug 3, 2022 at 10:15 PM Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> wrote:

Quote:
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> writes:

Quote:
Never mind, I found
https://freeswitch.org/confluence/display/FREESWITCH/verbose_sdp

Plus
https://freeswitch-users.freeswitch.narkive.com/jyFwqoeR/incompatible-destination-did-fs-not-offer-a-codec
--
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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brian at linuxpenguins...
Guest





PostPosted: Sun Aug 07, 2022 11:59 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Piotr Gregor <piotr@dataandsignal.com> writes:

Quote:
You can start from checking if these packets are sent at all, and if you
get those packets at all.
Use wireshark or tcpdump.
Do the same on both machines, you will see if and where RTP is sent/recved.
If all RTP is sent but it cannot reach your FreeSWITCH, then check ports
are allowed in the firewall.

But how do I know if the remote server which I don't have access to is
actually sending the RTP? I don't appear to see anything. This was a big
unknown. Is it sending? Is it sending to the correct address? etc.

I think I have it working now. Although somewhat confused, because I
tried this same setup earlier - or I thought I had - and it didn't work.

My theory:

* I could see outgoing RTP packets but not incoming RTP packets.

* Outboard RTP packets were being blocked by my firewall (EdgeRouter),
because its stateful connection tracking doesn't seem to be working
anymore.

* Remote server decided "seeing as I am not receiving any RTP packets I
am not going to bother sending any either". Actually this bit seems a
bit suspicious.

* By whitelisting 1024-65535 UDP outgoing (incoming was already
whitelisted), it works. Which isn't great solution, but maybe the best
I can do for now.

Result: It looks like incoming packets is the problem, but in actual
fact outgoing packets is the problem.

Also curious, when I run tcpdump on the EdgeRouter, it doesn't show any
of these UDP packets, even when there must be some because I have two
way audio communications. So the one really good tool I have to debug
firewall issues is actually very misleading.

There are some points in this theory that don't entirely add up (such as
the bandwidth graphs for the Internet port on the Internet side of the
firewall seemed to increase when blocked voice session in progress), but
I think the important part is that I do have this working again.
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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brian at linuxpenguins...
Guest





PostPosted: Mon Aug 08, 2022 7:02 am    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian May <brian@linuxpenguins.xyz> writes:

Quote:
But how do I know if the remote server which I don't have access to is
actually sending the RTP? I don't appear to see anything. This was a big
unknown. Is it sending? Is it sending to the correct address? etc.

Arrggh. I thought it was fixed, but now the problem has come back again.

I can see the outgoing audio packets, but I cannot see any incoming
audio packets.

Oh, OK need to use:

<action application="set" data="proxy_media=true"/>

Fair enough, the end phone is behind NAT.

But now while incoming audio is working, outgoing audio is not working.
Arrghhh!

tcpdump seems to show 3 streams:

* stream in from phone
* stream out to phone
* stream in from provider

But no stream out to provider.

How do I debug this?
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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brian at freeswitch.com
Guest





PostPosted: Mon Aug 08, 2022 7:34 am    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Proxy media there to allow things thru that FreeSWITCH doesn't know about the codecs, it's no longer useful and shouldn't be used as it isn't a fix for anything really.

/b




On Mon, Aug 8, 2022 at 6:58 AM Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> wrote:

Quote:
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> writes:

Quote:
But how do I know if the remote server which I don't have access to is
actually sending the RTP? I don't appear to see anything. This was a big
unknown. Is it sending? Is it sending to the correct address? etc.

Arrggh. I thought it was fixed, but now the problem has come back again.

I can see the outgoing audio packets, but I cannot see any incoming
audio packets.

Oh, OK need to use:

<action application="set" data="proxy_media=true"/>

Fair enough, the end phone is behind NAT.

But now while incoming audio is working, outgoing audio is not working.
Arrghhh!

tcpdump seems to show 3 streams:

* stream in from phone
* stream out to phone
* stream in from provider

But no stream out to provider.

How do I debug this?
--
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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brian at linuxpenguins...
Guest





PostPosted: Mon Aug 08, 2022 8:05 am    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Quote:
<action application="set" data="proxy_media=true"/>

Actually, I think I was getting confused here. I don't want that. But
simply:

<action application="set" data="bypass_media=false"/>

But that is the default anyway.

If I use bypass_media=false, then I get no inbound audio on inbound
calls. As in I don't see any packets. And the IP address we provide in
the SDP packet is correct. And I have firewall rules that explicitly
allow the traffic.

If I use proxy_media=true, then I get no outbound audio on inbound
calls. Which is weird that inboard audio works.

https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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brian at freeswitch.com
Guest





PostPosted: Mon Aug 08, 2022 12:36 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

you probably have a nat issue, what's the topology like? What is ext-*-ip set to, and what is local-network-acl set to?

On Mon, Aug 8, 2022 at 8:03 AM Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> wrote:

Quote:
> <action application="set" data="proxy_media=true"/>

Actually, I think I was getting confused here. I don't want that. But
simply:

<action application="set" data="bypass_media=false"/>

But that is the default anyway.

If I use bypass_media=false, then I get no inbound audio on inbound
calls. As in I don't see any packets. And the IP address we provide in
the SDP packet is correct. And I have firewall rules that explicitly
allow the traffic.

If I use proxy_media=true, then I get no outbound audio on inbound
calls. Which is weird that inboard audio works.

https://freeswitch.org/confluence/display/FREESWITCH/Proxy+Media
--
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com (sales@freeswitch.com)
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org (FreeSWITCH-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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brian at linuxpenguins...
Guest





PostPosted: Mon Aug 08, 2022 5:01 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian West <brian@freeswitch.com> writes:

Quote:
you probably have a nat issue, what's the topology like? What is ext-*-ip
set to, and what is local-network-acl set to?

provider (internet) <--firewall--> Freeswitch (external IP) <--firewall--> Phone (internal IP)

Freeswitch is outside the NAT, and looking at the packets, it is sending
the correct IP addresses to the provider.

All firewalls are whitelisted to send all UDP 1024-65535 through.

As far as freeswitch is concerned there is no NAT. Unless I try to get
the phone to talk directly to the provider. Which is something I am not
even attempting to do right now.

local-network-acl is set to localnet.auto

If there was a codec negotiation issue, what would the symptoms be? I
would expect dropped phone calls, not dropped audio. So I don't think
that is what I am experiencing here.

My gut feeling is that the remote provider is sending audio (wish I
could prove it) which means traffic is getting blocked somewhere - most
likely my firewall. So will concentrate my efforts here. Starting of by
disabling SIP support in my firewall.
https://community.ui.com/questions/VoIP-SIP-Phone-through-NAT-on-EdgeMax/0632ce0e-9b8d-4bf0-907f-ac55485e32e6
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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brian at freeswitch.com
Guest





PostPosted: Mon Aug 08, 2022 5:09 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Focus on the NAT issue between the phone and FreeSWITCH.  
/b





On Mon, Aug 8, 2022 at 4:47 PM Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> wrote:

Quote:
Brian West <brian@freeswitch.com (brian@freeswitch.com)> writes:

Quote:
you probably have a nat issue, what's the topology like? What is ext-*-ip
set to, and what is local-network-acl set to?

provider (internet) <--firewall--> Freeswitch (external IP) <--firewall--> Phone (internal IP)

Freeswitch is outside the NAT, and looking at the packets, it is sending
the correct IP addresses to the provider.

All firewalls are whitelisted to send all UDP 1024-65535 through.

As far as freeswitch is concerned there is no NAT. Unless I try to get
the phone to talk directly to the provider. Which is something I am not
even attempting to do right now.

local-network-acl is set to localnet.auto

If there was a codec negotiation issue, what would the symptoms be? I
would expect dropped phone calls, not dropped audio. So I don't think
that is what I am experiencing here.

My gut feeling is that the remote provider is sending audio (wish I
could prove it) which means traffic is getting blocked somewhere - most
likely my firewall. So will concentrate my efforts here. Starting of by
disabling SIP support in my firewall.
https://community.ui.com/questions/VoIP-SIP-Phone-through-NAT-on-EdgeMax/0632ce0e-9b8d-4bf0-907f-ac55485e32e6
--
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)>
https://linuxpenguins.xyz/brian/



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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brian at linuxpenguins...
Guest





PostPosted: Mon Aug 08, 2022 7:43 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian West <brian@freeswitch.com> writes:

Quote:
Focus on the NAT issue between the phone and FreeSWITCH.

All of that seems to be working...

Here is a partial tcpdump with packets to/from the server:

09:49:29.224492 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.243749 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.263882 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.283501 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.303796 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.321400 IP 59.167.180.194.19289 > 103.140.134.33.21005: UDP, length 112
09:49:29.323725 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.343927 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.363231 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.383833 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.403074 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.423905 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.443366 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.463924 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.483771 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.503919 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.523919 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.543937 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.548972 IP 103.140.134.33.21005 > 59.167.180.194.19289: UDP, length 84
09:49:29.563806 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.583645 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.604072 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.623863 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.643842 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.663620 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172
09:49:29.683939 IP 59.167.180.194.19288 > 103.140.134.33.21004: UDP, length 172

As you can see outgoing audio is getting sent. And it is confirmed to be
received, I can hear it.

But I simply am not getting the audio packets in.

Periodically I do get an RTCP packet in however. As in the 84 byte
packet above. This gets all the way through to my freeswitch server.

This RTCP is a sender report, and seems to indicate that the sender is
sending me packets, that the sender is using my correct IP address, and
no firewall is blocking any packet.

So how is it possible I am receiving RTCP sender reports, but I am not
receiving the data packets?
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
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brian at linuxpenguins...
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PostPosted: Mon Aug 08, 2022 9:33 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

OK, so some facts:

* Similar problems for 2 providers.
* If incoming call is bridged, it fails.
* If incoming call is redirected to 9196 echo test it works.
* The call to 9196 is answered immediately. But the time take to answer
the call does not appear to be significant.

I created tcpdumps of both cases.

* In both cases, simple INVITE, TRYING, OK, ACK sequence.
* In the good case, the provider starts sending data immediately after the ACK.

My reasoning is:

* The server must be doing something different in order for the
behaviour to change of the packets it is sending.

* This means the server must know which case is being tested.

* This follows that we must be communicating something different to the
server, depending on which case is being tested.

I have opened two instances of wireshark, and comparing the two traces,
I see only insignificant differences:

* port numbers are different.

* In the not-working case we send a Session Progress message while the
phone is ringing, I think this is just a consequence of taking more
time to answer the call.

* In the working case, in the OK message freeswitch sends a "Accept: application/sdp" header.
In the non-working case we omit the header.
I find it hard to believe this could be of any consequence.

All other details, codecs, IP address details, etc are identical.


Oh, almost missed a difference; we send to the provider in the OK
message (bad vs good calls):

-P-Asserted-Identity: "Outbound Call" <sip:1005@sip.crazytel.net.au>
+P-Asserted-Identity: "61390369013" <sip:61390369013@sip.crazytel.net.au>

What is this header? Could the fact I am sending the wrong value be
significant? My provider doesn't know anything about my 1005 extension.

I am somewhat doubtful actually.

--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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brian at freeswitch.com
Guest





PostPosted: Tue Aug 09, 2022 8:26 am    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Sounds like you need to setup outbound caller ID, do you have a full sip trace?

On Mon, Aug 8, 2022 at 9:22 PM Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)> wrote:

Quote:
OK, so some facts:

* Similar problems for 2 providers.
* If incoming call is bridged, it fails.
* If incoming call is redirected to 9196 echo test it works.
* The call to 9196 is answered immediately. But the time take to answer
  the call does not appear to be significant.

I created tcpdumps of both cases.

* In both cases, simple INVITE, TRYING, OK, ACK sequence.
* In the good case, the provider starts sending data immediately after the ACK.

My reasoning is:

* The server must be doing something different in order for the
  behaviour to change of the packets it is sending.

* This means the server must know which case is being tested.

* This follows that we must be communicating something different to the
  server, depending on which case is being tested.

I have opened two instances of wireshark, and comparing the two traces,
I see only insignificant differences:

* port numbers are different.

* In the not-working case we send a Session Progress message while the
  phone is ringing, I think this is just a consequence of taking more
  time to answer the call.

* In the working case, in the OK message freeswitch sends a "Accept: application/sdp" header.
  In the non-working case we omit the header.
  I find it hard to believe this could be of any consequence.

All other details, codecs, IP address details, etc are identical.


Oh, almost missed a difference; we send to the provider in the OK
message (bad vs good calls):

-P-Asserted-Identity: "Outbound Call" <sip:1005@sip.crazytel.net.au ([email]sip%3A1005@sip.crazytel.net.au[/email])>
+P-Asserted-Identity: "61390369013" <sip:61390369013@sip.crazytel.net.au ([email]sip%3A61390369013@sip.crazytel.net.au[/email])>

What is this header? Could the fact I am sending the wrong value be
significant? My provider doesn't know anything about my 1005 extension.

I am somewhat doubtful actually.

--
Brian May <brian@linuxpenguins.xyz (brian@linuxpenguins.xyz)>
https://linuxpenguins.xyz/brian/



--



Brian West | Co-founder and Developer
Need Commercial support? email sales@freeswitch.com (sales@freeswitch.com)
FreeSWITCH Solutions | 17345 Civic Drive #2531 Brookfield, WI 53045
Email: brian@freeswitch.com (brian@freeswitch.com)
Mobile: 918-424-9378
Website: https://www.FreeSWITCH.com
[/url] [url=https://twitter.com/freeswitch]
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brian at linuxpenguins...
Guest





PostPosted: Tue Aug 09, 2022 5:30 pm    Post subject: [Freeswitch-users] codec negotiation error Reply with quote

Brian West <brian@freeswitch.com> writes:

Quote:
Sounds like you need to setup outbound caller ID, do you have a full sip
trace?

Have a look at both traces here:

https://gist.github.com/brianmay/2ec0f404b901daf5a9e763aac0989cfe

I really thing the key to this problem must be somewhere here. Although
to my eyes everything looks OK.

Note there are two traces, one from my mobile to the audio repeat
service (9196) which worked, and one from my mobile to an extension
(1005) which didn't work.

Personally I find it hard to believe that sending the wrong ID in
response to the final OK message could cause loss of audio.

This isn't the outbound caller ID that was transmitted by the caller (my
provider) to the callee (freeswitch) on the invite, this is the callee
(freeswitch) telling my provider who they just called.

But if you think this could cause problems, and call tell me how to
change it (everything I see if for the outbound caller id), then I can
try to change it.
--
Brian May <brian@linuxpenguins.xyz>
https://linuxpenguins.xyz/brian/

_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com
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