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One way audio Internal and SIP IP Office 11.1...

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Feb 17, 2017
22
IN
Hi Guys, I have an setup of IP Office 11.1. and Grandstream open SIP Phones with latest firmware. I am facing issue with one way audio intercom and SIP trunk.
When i call from Phone 1 - 172.35.83.77 to Phone 2 - 172.35.85.203 or Phone 1 to Mobile No i am not able to hear on Phone 2 or Mobile. still able to hear from Phone 2 to Phone 1 or mob to phone 1.
I did some trace and found that call RTP are routed as below,
Phone 1 -- IP Office -- Undefined IP
172.35.83.77 --> 172.35.83.200 --> 172.35.85.98
Which need to be,
Phone 1 -- IP Office -- Phone 2
172.35.83.77 --> 172.35.83.200 --> 172.35.85.203

I am sharing the phone trace and SIP flow image can any one check the trace using wireshark and let me know who is misbehaving? Why the IP Office is sending RTP to another IP address rather than called phone IP. Only some phones has this issue and get resolved when phone takes another IP or assigned in static mode.
SIP_call_flow_berx39.png


[URL unfurl="true"]https://res.cloudinary.com/engineering-com/raw/upload/v1677918295/tips/out.pcap_na1kwd.gz[/url]
 
172.35.X.X is not private address space. You've most likely got a routing problem since you're using public IP addresses for your phones. (disregard if you own those public IPs and are using them for phones for some reason)
 
Hi, Ya actually client has provided the range. I find only some phones have the above issue i can say 4 to 5 out of 230 phone have issue were it gets resolved when i change the IP of that phone. I had set the default route as 172.35.0.0 255.255.0.0 and 172.35.83.1. where everything is working fine only have issue with some phones.
 
That address space needs to be changed. The problem will be solved when you aren't creating routes to public IPs on a private network.
 
Yep, make sure if they want to use 172.xxx.xxx.xxx that it's within the range of 172.16.0.0/12. Expressed in a generic range and not in CIDR that's 172.16.0.0 – 172.31.255.255
 
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