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Re: [Linphone-developers] ERROR Incompatible SDP answer received, need t


From: Russell Treleaven
Subject: Re: [Linphone-developers] ERROR Incompatible SDP answer received, need to abort the call
Date: Wed, 11 Apr 2018 16:41:39 -0400

the 200 ok seems to have come from 
2018-03-23 14:46:09:730 MESSAGE channel [0x5e32bb0]: received [1076] new bytes from [TCP://keyyo.net:5060]:

also

Server: PhoneSystems.net 00DA55B93751 Cisco/SPA504G-7.5.1a

They are the ones sending back the "a=ptime:30"



On Wed, Apr 11, 2018 at 2:58 PM, CircleCode <address@hidden> wrote:
sorry for the late answer, I waited to have a hand on the culprit phone…
I haven't found any way to configure ptime on it… :(

On Fri, Mar 23, 2018 at 9:30 PM, Russell Treleaven <address@hidden> wrote:
its a cisco.
cisco sucks.

On Fri, Mar 23, 2018 at 3:23 PM, Russell Treleaven <address@hidden> wrote:
Server: PhoneSystems.net 00DA55B93751 Cisco/SPA504G-7.5.1a
is setting setting ptime=30

If you control that peer try it without ptime set or set it to the sane value of 20

On Fri, Mar 23, 2018 at 11:07 AM, CircleCode <address@hidden> wrote:
I restarted linphone to be sure, then I double checked the config: ice is disabled (see https://framapic.org/639X2GLAEIQG/a4dpuLEDaFiV.png) But I can still see ICE and stun in the logs…
Can it be defined somewhere in some conf file, or could it be some kind of UPNP (mis)detection?

On Fri, Mar 23, 2018 at 3:52 PM, Russell Treleaven <address@hidden> wrote:
m=audio 7078 RTP/AVPF 0 8 101
a=rtpmap:101 telephone-event/8000
pcmu, pcma and rfc2883 dtmf

c=IN IP4 92.34.204.81
a=rtpmap:101 telephone-event/8000
a=candidate:1 1 UDP 2130706431 192.168.10.107 7078 typ host
a=candidate:1 2 UDP 2130706430 192.168.10.107 7079 typ host
a=candidate:2 1 UDP 1694498815 92.34.204.81 7078 typ srflx raddr 192.168.10.107 rport 7078
a=candidate:2 2 UDP 1694498814 92.34.204.81 7079 typ srflx raddr 192.168.10.107 rport 7079
stun and ice are still enabled and that stun that external ip address is inappropriate



On Fri, Mar 23, 2018 at 8:17 AM, Russell Treleaven <address@hidden> wrote:
This is what the peer agreed to

m=audio 42170 RTP/AVP 18 99 a=rtpmap:18 G729a/8000 a=fmtp:18 annexb=yes a=rtpmap:99 telephone-event/8000 a=fmtp:99 0-15 a=ptime:30 a=sendrecv

disable all codecs except pcmu and pcma
and try again
if it still fails try setting dtmf to sip info(this wont be a practical configuration but good for eliminating dtmf is the problem)
and try again
if it still fails disable ice
and try again
if it still fails switch to tcp
and try again
if it sill fails send another trace

good luck

On Fri, Mar 23, 2018 at 8:02 AM, CircleCode <address@hidden> wrote:
Thanks, I thought the log was not verbose enough by default.


I replaced my phone by `##_MINE_##` and my contact phone by `#_THEIRS_#`.
If there is anything else confidential in the logfile I would have missed, please tell me so that I can drop this version and upload a sanitized one.

On Thu, Mar 22, 2018 at 2:38 PM, Russell Treleaven <address@hidden> wrote:
The log file contains an inline sip trace. Clear the log. Do the test. Send the file

On Thu, Mar 22, 2018, 2:00 AM CircleCode <address@hidden> wrote:
let's be precise:
I'm using linphone-desktop on Arch linux, installed through flatpak.
About says:
Desktop 4.1.1 - Qt5.9.0
Core 3.12.0

Again, sorry for my lack of precisions :(

On Wed, Mar 21, 2018 at 8:23 PM, Russell Treleaven <address@hidden> wrote:
You should have been even more specific and said which version of Linphone. ;)

On Wed, Mar 21, 2018 at 2:54 PM, CircleCode <address@hidden> wrote:
Sorry, I should have been more specific: I'm using desktop version on Linux…

On Wed, Mar 21, 2018 at 7:52 PM, Russell Treleaven <address@hidden> wrote:
In the android client
configure settings->debug on
about-reset log

receive call

send log
choose yourself as a recipient

That log should contain a sip trace






On Wed, Mar 21, 2018 at 2:46 PM, CircleCode <address@hidden> wrote:
How can I get full sip trace?
the logs are not detailed enough to see SDP payload.

Thanks

On Wed, Mar 21, 2018 at 4:48 PM, Russell Treleaven <address@hidden> wrote:
Compare the sip trace of the successful call to the failing call, focus on the sdp payload of the offending request or response. Post the sip trace to the list if you need help.

On Wed, Mar 21, 2018, 11:40 AM CircleCode <address@hidden> wrote:
I'm using Linphone with a keyyo SIP account.

when I call people using a softphone, everything works correctly.
When I call someone with a physical sip phone, the call is aborted when my correspondent hang on.
When the same correspondent calls me from the same phone, everything works correctly.

In the logs, I get "ERROR Incompatible SDP answer received, need to abort the call" when the calls fails.

I do not find anything else interesting in the logs… anything I can investigate?
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--
Sincerely,

Russell Treleaven


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--
Sincerely,

Russell Treleaven


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--
Sincerely,

Russell Treleaven




--
Sincerely,

Russell Treleaven


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--
Sincerely,

Russell Treleaven




--
Sincerely,

Russell Treleaven


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--
Sincerely,

Russell Treleaven


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