Sip server decline Invite request if ICE enabled

[bensly]

1. Brekeke Product Name and Version: 3.1.1.2/340

2. Java version: 1.7.0_09

3. OS type and the version: Windows XP SP2

4. UA (phone), gateway or other hardware/software involved: IMSDROID

5. Your problem:

when make a call from imsdroid client which enabled ICE,sip server decline the request.kindly help me how to resolve this issue.if disable ICE works well

invite message follows.
INVITE sip:667@192.168.1.14:5060 SIP/2.0

Via: SIP/2.0/UDP 192.168.1.70:52333;branch=z9hG4bK543144397;rport

From: <sip:500@192.168.1.14:5060>;tag=1535017271

To: <sip:667@192.168.1.14:5060>

Contact: <sip:500@192.168.1.70:52333;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

Call-ID: 9dc9dd7d-0031-28d8-0712-60a119238501

CSeq: 1198677746 INVITE

Content-Type: application/sdp

Content-Length: 976

Max-Forwards: 70

Accept-Contact: *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"

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

Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE, REFER

Privacy: none

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

User-Agent: IM-client/OMA1.0 android-ngn-stack/v1.0 (doubango r687 - GT-S5360)

P-Preferred-Identity: <sip:500@59.90.246.89:5061>

Supported: 100rel



v=0

o=doubango 1983 678901 IN IP4 192.168.1.70

s=-

c=IN IP4 192.168.1.70

t=0 0

m=audio 41994 RTP/AVP 8 0 101

c=IN IP4 223.177.170.155

a=ptime:20

a=silenceSupp:off - - - -

a=rtpmap:8 PCMA/8000/1

a=rtpmap:0 PCMU/8000/1

a=rtpmap:101 telephone-event/8000/1

a=fmtp:101 0-16

a=tcap:1 RTP/AVPF

a=pcfg:1 t=1

a=sendrecv

a=rtcp-mux

a=ssrc:3253488559 cname:ldjWoB60jbyQlR6e

a=ssrc:3253488559 mslabel:6994f7d1-6ce9-4fbd-acfd-84e5131ca2e2

a=ssrc:3253488559 label:Doubango

a=ice-ufrag:SVDKL1f5KI1be4D

a=ice-pwd:AwvHnazU6CumzUTN1njjw

a=mid:audio

a=candidate:hKhKC9VcB 1 udp 2130706175 223.177.170.155 41994 typ host

a=candidate:hKhKC9VcB 2 udp 2130706174 223.177.170.155 41995 typ host

a=candidate:eXE9YFUmB 1 udp 2130705919 192.168.1.70 25612 typ host

a=candidate:eXE9YFUmB 2 udp 2130705918 192.168.1.70 25613 typ host

a=candidate:srflxeXE9 2 udp 1694498814 122.183.253.206 22095 typ srflx

a=candidate:srflxeXE9 1 udp 1694498815 122.183.253.206 22107 typ srflx


regards
bensly

[Laurie]

Are both SIP Server and SIP UA located in the same LAN?

Which SIP response code did the SIP Server return for declining?

[bensly]

Yes sip server and SIP UA located in same LAN.

Received response code is 603.

[Laurie]

If you use both SIP UA and SIP Server in the same LAN, you must disable NAT traversal feature such as ICE at the SIP UA because the SIP Server doesn't know the global IP address.

Otherwise, you need to add the same global IP address at the SIP Server's setting as an interface IP.
[Configuration] -> [System ] page -> [Network] -> [Interface address]

[bensly]

i have added the public ip address in the interface address.but still it decline the request.previously it was works.

[Laurie]

Is your public IP address changeable dynamically? (e.g. DHCP..)


If your router supports UPnP, let you enable it at Brekeke SIP Server.
With the UPnP, the SIP Server can obtain its public IP address automatically.

You can find the UPnP setting at the [Configuration]->[System] page.

http://wiki.brekeke.com/wiki/UPnP
http://wiki.brekeke.com/wiki/Configuration-for-BSS-behind-NAT


But.. in your situation, I recommend that you disable ICE at the SIP client.

[bensly]

we are using static ip address only.
we are developed mobile based app and given to the client.so client does't want to know and change the IP address and NAT info.so we are used static public IP address to register with sip.so clients may be in-house or out-house.we wants the client to work in both scenario without changing the settings. Thats the reason we are using public IP and ICE.

[Laurie]

Try the following DialPlan rule.

Matching Patterns

$request = ^INVITE To = sip:(.+)@

Deploy Patterns

To = sip:%1@

Which SIP SDK are you using??

[bensly]

i am using imsdroid app in android mobile.

[bensly]

After deployed the dial plan also the call declined. if mobile in 3G network also the call declined.if i enable ICE, all the scenario the call was declined.

[Laurie]

Do you want to make a call between registered users?
Can you see any error phrases in the SIP Server's [Logs] page with the response code 603?


Let you paste all of packets here.

[bensly]

yes i want to make call between registered users.

and as per the log it says "Packet Too Big"

log is:
7, sip:666@192.168.1.237, sip:125@192.168.1.237, 0, 1357543153468, , 1357543153453, Packet Too Big, 603, 192.168.1.78:51854, 192.168.1.237:49152, 0,

[Laurie]

> and as per the log it says "Packet Too Big"

It is the reason of the issue! Why did you not check the log page at the first??


The default MTU size is 1500byte.
SIP packet sent over UDP must follow this size limitation.
http://en.wikipedia.org/wiki/Maximum_transmission_unit


There are several ways to avoid the issue.

- Reduce your SIP packet size
- Use TCP instead of UDP
- Tune the MTU size with the "net.sip.mtu.size" in the [Configuration]->[Advanced] page.
For example:
net.sip.mtu.size = 2000

[bensly]

Thanks for your great help, and its works well now.

[Laurie]

Glad to know it!