Thanks for the information.if i enable ICE the packet size goes beyond MTU limit size 1500 bytes.so call landing issues coming in low bandwith network.so to avoid that i have disabled that option.can you let me know how to instruct to sip server to use 208.54.xxx.xxx for signaling and RTP, if ip ...

i am using imsdroid client. i have disabled stun and ice. but sip server behive correctly and works if private and public ip address like this 10.35.xxx.xxx 112.79.xxx.xxx but i don't know why sip server behive differently if ip address like this 162.176.xxx.xxx 208.54.xxx.xxx

i have tried with the dial plan,but still its using the private ip address for RTP and INVITE.

1. Brekeke Product Name and Version: brekeke sip server 3.1.7.8/348.2 2. Java version: 1.7.0_21 3. OS type and the version: Windows 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: i have a sip client loaded in android mobile.i am using 3G network to register the mobile ...

Hi,
Thanks.its working now.

Regards
Bensly

1. Brekeke Product Name and Version: 2. Java version: 3. OS type and the version: 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: i have deployed two dial plan below 1 NAT IP change $request = ^INVITE $ifsrc = 59.90.xx.xx $ifdst = 59.90.xx.xx 2 PBX forwarding $request ...

if i use dial plan &net.rtp.ifsrc = 59.90.246.89 ,&net.rtp.ifdst = 59.90.246.89 like this, still it does't replace to global ip address. For RTP/AVPF problem already i have pasted the SIP trace in above conversation.in that you can find invite will have RTP/AVP and response will be RTP/AVPF. i have ...

i have checked the same as you suggested instead of advance page,if i put dial plan like this $request = ^INVITE $ifsrc = 59.90.246.89 $ifdst = 59.90.246.89 public IP replaced,and audio works both side now. and i am facing one more problem in audio. if i make a call A - > B audio works. if i call B ...

I have tried as you suggested below in advance page

net.rtp.ifsrc=59.90.246.89
net.rtp.ifdst=59.90.246.89


but still it does't replace to public IP address.

>> Are SIP Server and SIP clients located in the same LAN behind NAT? yes, sip clients and sip server are located in same LAN behind NAT. we are registering the client using public IP. because the clients are running in andorid mobile.so the user may use local WIFI or GPRS network.so this IP address ...

The IP address as you said is correct. and in Interface also listed my local and public IP. for more information i have enabled the sip server logs. please find the log below to get identify the problem. ================================================================================ Brekeke SIP ...

Sip server behind NAT,and i have configured local and public IP in interface 1 and 2. and RTP replay is ON , RTP Relay ( UA) is auto.port mapping is source port, and send UA remote address is yes. and after configured this settings i restarted the sip server also.it not replacing the public IP in ...

Hi, [Port Mapping] settings selected as source port only.and RTP relay mode is on.still same problem.client using local IP address in the SDP,but sip server not replacing the public IP address. i have tested with asterisk server.in that case all the call audio flows both side and works well.i don't ...

Hi,
i am using imsdorid client in android mobile.as you said in SDP local ip address only sending by the client.but after enabled Send UA's remote address= yes also the problem is same.

Regards
Bensly

1. Brekeke Product Name and Version: v3.1.7.8 2. Java version: 3. OS type and the version: Windows 2003 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: i have two IP client running on android mobile.both are registered successfully using public IP address.and both ...

Hi,
i have upgraded to v3.1.7.8. but still the issue there.i am creating new post for this.please replay on the new post.

Regards
Bensly

Hi, Thanks for the solutions and its worked for me.but now i am facing another problem is both side no audio.but call successfully established. i am able to see the RTP packet goes to sip server from two end points.but sip server not delivering the RTP to the clients.kindly let me know how to solve ...

Hi laurie.
Thanks for your suggestion. i will check the setting as you said.and let you know the status shortly.

Regards
Bensly

1. Brekeke Product Name and Version: 2. Java version: 1.5.0_06 3. OS type and the version: Linux 2.6.9-89.ELsmp 4. UA (phone), gateway or other hardware/software involved: 5. Your problem: i have two sip client installed in my mobile.both are behind NAT. i have configured Public IP address in sip ...

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

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,

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.

i am using imsdroid app in android mobile.

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 ...

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