[Freeswitch-users] ICE issue

Bastien LE REST ba.lerest at gmail.com
Mon Jun 13 22:26:24 MSD 2016


Hi Victor,

I think the problem is the transcoding. Maybe the iOS call works fine
because there is the codec G722 available on both legs.

Please find my FS configuration here. The configuration is from scratch.
All other parameters value are set by default.

FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad
2016-05-05 15:38:32Z 64bit)

*vars.xml*

<X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,speex at 8000h
> @20i,speex at 16000h@20i,speex at 32000h@20i"/>

<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=G722"/>


*internal.xml*

<param name="inbound-codec-prefs" value="$${global_codec_prefs}"/>
> <param name="outbound-codec-prefs" value="$${outbound_codec_prefs}"/>

<param name="inbound-late-negotiation" value="false"/>
> <param name="inbound-zrtp-passthru" value="false"/>
> <param name="disable-transcoding" value="false"/>
> <param name="inherit_codec" value="false"/>
> <param name="bypass_media" value="false"/>
> <param name="proxy_media" value="false"/>


*dialplan*

<action application="bridge"
> data="{media_mix_inbound_outbound_codecs=true,absolute_codec_string='G722'}sofia/gateway/
> sip.nl.didlogic.net/${destination_number}"/>


*show codec *

codec,G.722,mod_spandsp
> codec,Speex,CORE_SPEEX_MODULE
> codec,iLBC,mod_ilbc


*sofia status profile internal*

Name             internal
> Domain Name       N/A
> Auto-NAT         false
> DBName           sofia_reg_internal
> Pres Hosts       188.166.40.202,188.166.40.202
> Dialplan         XML
> Context           public
> Challenge Realm   auto_from
> RTP-IP           188.166.40.202
> SIP-IP           188.166.40.202
> URL               sip:mod_sofia at 188.166.40.202:5060
> BIND-URL         sip:mod_sofia at 188.166.40.202:5060;transport=udp,tcp
> WS-BIND-URL     sip:mod_sofia at 188.166.40.202:5066;transport=ws
> WSS-BIND-URL     sips:mod_sofia at 188.166.40.202:7443;transport=wss
> HOLD-MUSIC       local_stream://moh
> OUTBOUND-PROXY   N/A
> CODECS IN       OPUS,speex at 8000h@20i,speex at 16000h@20i,speex at 32000h@20i
> CODECS OUT                 G722
> TEL-EVENT                     101
> DTMF-MODE                   rfc2833
> CNG                                13
> SESSION-TO                  0
> MAX-DIALOG                 0
> NOMEDIA                       false
> LATE-NEG                      false
> PROXY-MEDIA               false
> ZRTP-PASSTHRU          false
> AGGRESSIVENAT         false
> CALLS-IN                       1
> FAILED-CALLS-IN         1
> CALLS-OUT                   0
> FAILED-CALLS-OUT     0
> REGISTRATIONS          1


Thank you for you help.

Kind regards,
Bastien.

On 13 June 2016 at 18:31, Victor Medina <vittico at gmail.com> wrote:

> In vars.XML. whats in your inbound and outbound códec list?
> El 12/6/2016 6:26, "Bastien LE REST" <ba.lerest at gmail.com> escribió:
>
>> Hello Victor,
>>
>> Yes, I configured FS to only use iLBC codec with mod_iLBC. iOS works fine
>> using this codec but I still have the same problem with android.
>>
>> Here's the trace: https://pastebin.freeswitch.org/view/fcf3b334
>>
>> Thank you for your help, it's much appreciated.
>>
>> Kind regards,
>> Bastien.
>>
>> On 11 June 2016 at 21:22, Victor Medina <vittico at gmail.com> wrote:
>>
>>> Are they enabled con FS side??
>>> El 11/6/2016 5:39, "Bastien LE REST" <ba.lerest at gmail.com> escribió:
>>>
>>>> Hello Victor,
>>>>
>>>> Thank you for your reply. I tried to add both iLBC and Speex as a
>>>> priority codec in the codecs list but unfortunately that didn't fix the
>>>> problem.
>>>>
>>>> Is it normal that FS doesn't find a candidate for RTCP?
>>>>
>>>> Kind regards,
>>>> Bastien.
>>>>
>>>> On 10 June 2016 at 19:33, Victor Medina <vittico at gmail.com> wrote:
>>>>
>>>>> More...
>>>>>
>>>>> On iOS you seem to be sending:
>>>>>
>>>>>
>>>>>    1.    a=rtpmap:98 speex/16000
>>>>>    2.    a=rtpmap:97 speex/8000
>>>>>    3.    a=rtpmap:99 speex/32000
>>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>>    5.    a=fmtp:104 mode=30
>>>>>    6.    a=rtpmap:3 GSM/8000
>>>>>    7.    a=rtpmap:0 PCMU/8000
>>>>>    8.    a=rtpmap:8 PCMA/8000
>>>>>    9.    a=rtpmap:9 G722/8000
>>>>>
>>>>>
>>>>> While on Android I can only see:
>>>>>
>>>>>
>>>>>    1.    a=rtpmap:98 speex/16000
>>>>>    2.    a=rtpmap:97 speex/8000
>>>>>    3.    a=rtpmap:99 speex/32000
>>>>>    4.    a=rtpmap:104 iLBC/8000
>>>>>
>>>>>
>>>>> FS is correctly complaining about it:
>>>>>
>>>>> 2016-06-10 12:48:33.533730 [ERR] mod_sofia.c:2346 CODEC NEGOTIATION
>>>>> ERROR.  SDP:
>>>>>
>>>>>
>>>>> Sin mas a que hacer referencia,
>>>>>
>>>>> Victor Medina
>>>>>
>>>>> On Fri, Jun 10, 2016 at 2:26 PM, Victor Medina <vittico at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> There is seems to be a codec issue.
>>>>>> Are you sure codecs are right on the Android side?
>>>>>>
>>>>>>
>>>>>> Sin mas a que hacer referencia,
>>>>>>
>>>>>> Victor Medina
>>>>>>
>>>>>> On Fri, Jun 10, 2016 at 1:28 PM, Bastien LE REST <ba.lerest at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Bonjour,
>>>>>>>
>>>>>>> I thought the issue was solved since the calls work just fine now
>>>>>>> with PJSIP on iOS, however the calls don't work with PJSIP on Android and I
>>>>>>> don't understand why, the PJSIP configuration is exactly the same on both
>>>>>>> devices.
>>>>>>>
>>>>>>> FS finds a candidate for RTP but it doesn't find a candidate for
>>>>>>> RTCP. Please note that I only have 3 candidate.
>>>>>>>
>>>>>>> FS trace: https://pastebin.freeswitch.org/view/8d4df983
>>>>>>>
>>>>>>> Could you please have a look at the trace, and let me know what am I
>>>>>>> doing wrong?
>>>>>>>
>>>>>>> Thank you very much in advance for your help in this matter.
>>>>>>>
>>>>>>> Kind regards,
>>>>>>>
>>>>>>> Bastien.
>>>>>>>
>>>>>>> On 4 June 2016 at 13:44, Bastien LE REST <ba.lerest at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi,
>>>>>>>>
>>>>>>>> When I enable ICE using PJSIP, FreeSwitch answers "488 Not
>>>>>>>> Acceptable Here". When I disable ICE, I don't have any problems, the calls
>>>>>>>> work just fine. I tried to use another SIP client with ICE and the call
>>>>>>>> works fine. So I think the problem has to do with PJSIP.
>>>>>>>>
>>>>>>>> I'm working on the migration from an Asterisk architecture to a
>>>>>>>> FreeSwitch architecture. I have to deal with PJSIP and ICE support because
>>>>>>>> my client is a mobile application who is already published on the stores.
>>>>>>>>
>>>>>>>> PJSIP trace : https://pastebin.freeswitch.org/view/ddeaae4b
>>>>>>>> FreeSwitch trace: https://pastebin.freeswitch.org/view/d380e01d
>>>>>>>>
>>>>>>>> Could you please have a look at the trace and tell me what am I
>>>>>>>> doing wrong? Thank you in advance.
>>>>>>>>
>>>>>>>> Kind regards,
>>>>>>>> Bastien.
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
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>>>>>>
>>>>>>
>>>>>
>>>>>
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>>>>
>>>>
>>>>
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>>
>>
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>> http://www.freeswitchsolutions.com
>>
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>>
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>
> _________________________________________________________________________
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>
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