Android imsdroid to PSTN call is not working

Question

i am using FreeSWITCH server and integrated with twilio SIP turnk. i am using android imsdroid application for making sip calls. imsdroid to imsdroid call is happening. imsdroid to PSTN no (i.e mobile number) call is not working. Gateway timeout error is shown in imsdroid. INVITE is sent to FreeSWITCH and freeswitch server is routing the call to Twilio. But not receiving any response. What could be the issue.

IP is allowed.From SIPML5 client to PSTN call is working. Below is the INVITE that is going from FreeSWITCH

INVITE sip:+919986790176@nowconf.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bK5yt865UUKDF4H
Max-Forwards: 69
From: "+919845217138" <sip:FreeSWITCH@nowconf.pstn.twilio.com>;tag=eFNKtBae4HN1r
To: <sip:+919986790176@nowconf.pstn.twilio.com>
Call-ID: 06f7df5f-1f81-1235-67b0-2e81eca04f81
CSeq: 98923179 INVITE
Contact: <sip:gw+Twilio-outbound@220.227.38.107:5080;transport=udp;gw=Twilio-outbound>
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
X-FS-Support: update_display,send_info
Remote-Party-ID: "+919845217138" <sip:+919845217138@nowconf.pstn.twilio.com>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1478495490 1478495491 IN IP4 220.227.38.107
s=FreeSWITCH
c=IN IP4 220.227.38.107
t=0 0
m=audio 22996 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

SIPML5 INVITE

INVITE sip:+919742164769@nowconf.pstn.twilio.com SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bKD5jv3tcpNtFDS
Max-Forwards: 69
From: "+919845217138" <sip:FreeSWITCH@nowconf.pstn.twilio.com>;tag=cvS8XKDp62Z5e
To: <sip:+919742164769@nowconf.pstn.twilio.com>
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 98788399 INVITE
Contact: <sip:gw+Twilio-outbound@220.227.38.107:5080;transport=udp;gw=Twilio-outbound>
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 488
X-FS-Support: update_display,send_info
Remote-Party-ID: "+919845217138" <sip:+919845217138@nowconf.pstn.twilio.com>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1478224523 1478224524 IN IP4 220.227.38.107
s=FreeSWITCH
c=IN IP4 220.227.38.107
t=0 0
m=audio 24404 RTP/AVP 102 9 0 8 103 101
a=rtpmap:102 opus/48000/2
a=fmtp:102 useinbandfec=0; cbr=1; maxaveragebitrate=30000; maxplaybackrate=48000; ptime=20; minptime=10; maxptime=40; stereo=1
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:103 telephone-event/48000
a=fmtp:103 0-16
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
ACK sip:172.18.22.113:5060 SIP/2.0
Via: SIP/2.0/UDP 220.227.38.107:5080;rport;branch=z9hG4bKeecN5NXSj35Zm
Route: <sip:54.172.60.3:5060;lr;ftag=cvS8XKDp62Z5e>
Max-Forwards: 70
From: "+919845217138" <sip:FreeSWITCH@nowconf.pstn.twilio.com>;tag=cvS8XKDp62Z5e
To: <sip:+919742164769@nowconf.pstn.twilio.com>;tag=70832428_6772d868_27e46324-e125-4895-a6ff-98031cdb43fc
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 98788399 ACK
Contact: <sip:gw+Twilio-outbound@220.227.38.107:5080;transport=udp;gw=Twilio-outbound>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.172.60.3:5060;branch=z9hG4bK763a.52f422d4.0
Via: SIP/2.0/UDP 172.18.22.113:5060;rport=5060;received=54.172.61.235;branch=z9hG4bK27e46324-e125-4895-a6ff-98031cdb43fc_6772d868_294770755291215
From: <sip:+919742164769@nowconf.pstn.twilio.com>;tag=70832428_6772d868_27e46324-e125-4895-a6ff-98031cdb43fc
To: "+919845217138" <sip:FreeSWITCH@nowconf.pstn.twilio.com>;tag=cvS8XKDp62Z5e
Call-ID: 69171aa5-1d0d-1235-67b0-2e81eca04f81
CSeq: 1 BYE
User-Agent: FreeSWITCH-mod_sofia/1.6.10+git~20160824T215404Z~726448d962~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
Supported: timer, path, replaces
Content-Length: 0

please post the siptrace here.you have to add your ip to twilio so it can allow call from it — suren, Nov 08 '16 at 05:53
if you will post the siptrace then easily i can tell you the what the main reason behind it. — suren, Nov 08 '16 at 05:55
IP is allowed.From SIPML5 client to PSTN call is working. Below is the INVITE that is going from FreeSWITCH — Hansika Penugonda, Nov 08 '16 at 06:16
invite packet seems fine.will you confirm that on this ip: 220.227.38.107 only you are checking weather you are getting response or not — suren, Nov 08 '16 at 06:53
please post both of invite .invite from sipml5 and freeswitch. — suren, Nov 08 '16 at 06:54
The first invite request is relate to imsdroid only. From freeswitch server invite is sending to twilio — Hansika Penugonda, Nov 08 '16 at 07:21
sdp part is not there in the first one.please send that too. — suren, Nov 08 '16 at 07:53
P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000.please remove these two header before sending to twilio — suren, Nov 08 '16 at 09:48
please go threw this ..https://wiki.freeswitch.org/wiki/Strip_SIP_Headers — suren, Nov 08 '16 at 10:25
for removing header you to do this before bridge call to pstn twilio sip gateway. — suren, Nov 08 '16 at 10:31

score 2 · Answer 1 · edited Nov 11 '16 at 11:50

2

Please remove these two headers before sending to Twilio:

P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel 

P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000

To do so you have to write these two line before sending call to Twilio sip gateway:

<action application="unset" data="sip_h_P-Preferred-Service"/> 

<action application="unset" data="sip_h_P-Access-Network-Info"/>

edited Nov 11 '16 at 11:50

halfer

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answered Nov 08 '16 at 11:48

suren

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1

Call established.But voice is audible only one side.On PSTN(i.e mobile) i am able to hear voice. But on imsdroid there is no voice. I took trace on freeswitch RTP packets are coming from Twilio , but freeswitch is not giving it to android client – Hansika Penugonda Nov 09 '16 at 11:11
please check firewall on freeswitch also add rule for the rtp port on firewall. – suren Nov 09 '16 at 11:54

Android imsdroid to PSTN call is not working

1 Answers1